| 00:00.02 | *** join/#asterisk [hC] (~hardcore@c-69-180-109-192.hsd1.fl.comcast.net) |
| 00:00.43 | Nethab | a specific codec? |
| 00:01.09 | *** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 00:01.50 | [hC] | Im having some bizarre iax trunking problems... I'm using asterisk from cvs head this month, and connecting to a guy using cvs head from about 8 months ago.. he can connect to me and send calls to me, but I cannot connect to him and send calls to him. I've verified the iax.conf and the dial command over and over, and it all looks correct. I get the error ""No Authority found" - Is there any way to debug this further to see why i'd b |
| 00:01.50 | [hC] | e getting it? |
| 00:02.07 | jontow | yes, any one.. are there recommendations to try? |
| 00:02.26 | shido6 | pastebin.ca your dialplan |
| 00:02.29 | shido6 | err |
| 00:02.31 | shido6 | iax.conf |
| 00:02.32 | shido6 | rather |
| 00:02.49 | Nethab | i thought trunking in HEAD was buggy now |
| 00:03.14 | tzanger | Nethab: there are a bunch of patches poised to go in to fix it with the new jb |
| 00:03.34 | Nethab | as soon as kpfleming starts |
| 00:03.36 | Nethab | ? |
| 00:03.45 | [hC] | Im using head from march i believe |
| 00:06.44 | jontow | i think ill try ilbc |
| 00:07.02 | jontow | looks pretty unintensive as far as low bandwidth goes |
| 00:07.06 | jontow | sure beats the hell out of ulaw :) |
| 00:07.36 | Nethab | quality suffers but bandwidth is saved |
| 00:07.58 | jontow | thats fine |
| 00:08.08 | jontow | it sounded pretty good over the IAX2 trunk i just tested on |
| 00:08.11 | *** join/#asterisk TonyAlmeida (~tonyalmei@61.33.161.6) |
| 00:08.25 | jontow | i mean, compared to jittery scratched broken nasty ulaw over low bandwidth links, it sounded GREAT :) |
| 00:09.43 | [hC] | ill paste these configs to pastebin here in a sec, is there something other than iax2 debug i can use to try to narrow down what would be causing the no authority to come up? |
| 00:10.47 | jontow | actually.. it sounds good even trunked over IAX2 then over the PRI :) not bad at all :))) i think ill give that a shot when i get home |
| 00:11.01 | shido6 | yes |
| 00:11.04 | shido6 | dialplan |
| 00:11.11 | shido6 | paste your dialplan an iax.conf |
| 00:11.35 | shido6 | im headin out to timmy's to get some fewd so I'll be back in 20 minutes |
| 00:12.26 | syle | how do phone numbers work exactly , how do you port them into voip internet space from PSTN? |
| 00:14.08 | *** join/#asterisk anthm (~anthm@h46085691.area4.spcsdns.net) |
| 00:14.08 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:14.50 | jontow | holy crap, it almost works |
| 00:14.51 | jontow | hahaha |
| 00:15.03 | [hC] | here we go, here's my pastebin |
| 00:15.04 | [hC] | http://pastebin.ca/11632 |
| 00:17.25 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 00:18.05 | Nethab | your using iax v1 |
| 00:18.09 | Nethab | ? |
| 00:18.11 | [hC] | no, iax2. |
| 00:18.32 | [hC] | My dial states IAX2/ and the debug states VERSION: 2 |
| 00:19.05 | Nethab | your port is 5036, isn't that the old iax1 port |
| 00:21.15 | *** join/#asterisk shmaltz (~chatzilla@ool-43551098.dyn.optonline.net) |
| 00:21.20 | shmaltz | hi everyone |
| 00:22.12 | shmaltz | is there a way to have queues ring sip phones with a strategy based on who is on the phone (even if they are just answering direct or making outgoing calls)? |
| 00:26.20 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 00:27.14 | |Vulture| | anyone know why I might be getting "Out of g.729 Decoder Licenses" errors, I have 2 licenses and 1 7960 is connected happens only when I try to bridge the call |
| 00:28.11 | vpp | hmm |
| 00:28.21 | vpp | i setup username=blah, host=someip |
| 00:28.27 | |Vulture| | it says: 1/2 encoders/decoders of 2 licensed channels are currently in use |
| 00:28.46 | vpp | now how do i identify all calls that come in with that username and from that host in extensions.conf? |
| 00:30.26 | jontow | goddamned impressive.. i can't believe i passed intelligible sound over a dialup with VoIP :P |
| 00:30.49 | Nethab | with ilbc? |
| 00:35.49 | *** join/#asterisk nwhit (~nwhit@64.50.35.78) |
| 00:36.46 | nwhit | hey guys... i am having a little problem with a t-1 using fxs_ks signalling.... asterisk doesn't always detect that the remote side picks up the phone... any ideas? |
| 00:37.06 | nwhit | or it takes a while for it to detect it |
| 00:37.23 | [hC] | shido6: still out? :) |
| 00:37.40 | outtolunc | nwhit: make sure that [context] has a exten t |
| 00:38.02 | outtolunc | otherwise it can take like 5-8 secs for it to fallthru |
| 00:38.09 | nwhit | ok |
| 00:38.17 | nwhit | i'll try that |
| 00:39.14 | shmaltz | is there a way to have queues ring sip phones with a strategy based on who is on the phone (even if they are just answering direct or making outgoing calls)? |
| 00:39.34 | newmedian | Off Topic: I'm looking for an IRC (client/)logger for Linux that will doggedly reconnect (if network problems) and log one or more channels to plaintext files. Any recommendations? (Prefer a non-GUI client/logger) |
| 00:41.23 | nwhit | outtolunc, what should it look like... exten => t,??? |
| 00:41.39 | outtolunc | exten => t,1,Hangup |
| 00:42.41 | outtolunc | years ago i had that issue and everyone said 'use kewlstart' ... well making sure you dialplan is correct is a good start <G> |
| 00:43.03 | outtolunc | it was hard for me, seeing as how my chanbank didn't support it <G> |
| 00:43.24 | outtolunc | it's set to loopstart and with a exten t works great |
| 00:45.58 | nwhit | outtolunc, that didn't seem to fix it... when I make a call out from, it goes Zap/1-1 Ringing .... and sometimes never reports Zap/1-1 answered SIP/201 |
| 00:46.06 | outtolunc | side note: you should also have an h |
| 00:46.15 | nwhit | i have an h and i and t |
| 00:46.33 | outtolunc | you are probably use a dial .... 20|r |
| 00:46.38 | outtolunc | or similar |
| 00:46.40 | Nethab | what about the s |
| 00:46.45 | Nethab | s h i t |
| 00:46.50 | nwhit | haha |
| 00:47.07 | outtolunc | s is only needed IF you send to that context without an exten# |
| 00:47.08 | *** join/#asterisk geekguy (~cdra@203.221.214.48) |
| 00:47.14 | Nethab | s = default, h = hangup, i = invalid, t = timeout |
| 00:47.55 | outtolunc | s isn't default perse, it's something got here without and exten.. so use me |
| 00:47.55 | nwhit | Dial(Zap/g1/${EXTEN}) |
| 00:47.58 | geekguy | Hi Peoples |
| 00:47.59 | ariel_ | (S) tart (H) angup (I) nvalid (T) imeout. hummm never |
| 00:48.33 | geekguy | does anyone have info on a OCTTEL SP4220 please |
| 00:48.41 | ariel_ | how things get put to gether is really strange. |
| 00:48.42 | nwhit | it works perfect sip to sip and zap incoming to sip |
| 00:48.51 | outtolunc | if you design your dialplan, it's 'rare' you ever need an s |
| 00:49.33 | outtolunc | just remember: for it to 'fallthru' to s, takes time |
| 00:49.49 | ariel_ | Not if you start it there. |
| 00:49.50 | outtolunc | that is unless you SENT it to s |
| 00:50.01 | [hC] | http://pastebin.ca/11632 - Any ideas why i might be seeing this error/ |
| 00:50.07 | newmedian | Wasn't S the letter that the guy on Sesame Street was trying to sell in the alley? They stopped showing that segment, of course, Don't want to encourage drugs or luring kids into alleys. |
| 00:50.12 | [hC] | Peer side is cvs head from 8-9 mos ago, client side is cvs head from last month |
| 00:50.17 | newmedian | Pssst. anyone want to buy the letter S? |
| 00:50.31 | Nethab | I thought you got r-s-t-l-n-e for free |
| 00:51.31 | nwhit | any ideas on my problem? |
| 00:52.21 | outtolunc | AB1 by chance? |
| 00:52.29 | ariel_ | exten => _6048803451,1,Dial(IAX2/${EXTEN}@dolphintel) |
| 00:52.38 | jontow | with iLBC and GSM |
| 00:52.44 | vpp | nwhit whats the problem? |
| 00:52.52 | jontow | though they're both quite scratchy, and i think a well-tuned speex might be my real solution |
| 00:53.32 | ariel_ | [hC], did you get that dial string. |
| 00:53.35 | outtolunc | nwhit: you said, zap, you didn't say what hardware was in the mix |
| 00:53.49 | nwhit | vpp, when i dial out from my sip phone through a zap channel (which is on a t-1 fxs_ks) the call takes a long time for asterisk to recognize it going from ringing to connected |
| 00:53.58 | nwhit | sangoma wanpipe |
| 00:54.00 | outtolunc | there are some chanbanks that just can't handle disconnect supervision |
| 00:54.05 | nwhit | robbed bit t-1 |
| 00:54.09 | vpp | nwhit i'm using the same hardware |
| 00:54.20 | vpp | just this second configured the exact same thing lol |
| 00:54.38 | vpp | except i'm using PRI |
| 00:54.52 | nwhit | vpp, i have great luck with them for pri, data, etc.... i don't know if it is the card or the wanpipe drivers |
| 00:55.10 | [hC] | ariel_: you want me to try dialing like that? |
| 00:55.15 | vpp | i just switched to the beta7 one.. had problems with the 2.3.3 |
| 00:55.23 | vpp | how long is 'long' ? |
| 00:55.38 | vpp | 2 or 3 seconds? |
| 00:55.39 | nwhit | i was using loopstart because that is what the telco told me to use, but it wasn't detecting remote hangups... now with kewstart it is, but not the connections |
| 00:55.41 | vpp | 10 minutes? |
| 00:55.42 | vpp | heheh |
| 00:56.02 | nwhit | 5secs --- 1min --- never.... it really varies on each call |
| 00:56.20 | vpp | channels bank? |
| 00:57.26 | nwhit | adit 600 |
| 00:57.50 | ariel_ | [hC], yes |
| 00:58.19 | nwhit | i just got a fix... i set turned off callprogress in zapata.conf and it immediately goes to a connected state when it dials... thats what I want |
| 00:58.22 | nwhit | fixed |
| 00:58.25 | nwhit | :) |
| 00:59.09 | [hC] | ariel_: it didnt work |
| 01:00.47 | outtolunc | but without callprogress=yes, on an e&m line don't you lose 'busy's? |
| 01:01.17 | [hC] | did fwd stop allowing people to dial toll free numbers thru their free service? |
| 01:02.11 | [hC] | nevermind. |
| 01:03.40 | ariel_ | [hC], you can also dial via exten => _X.,dial,iax2/username:password@ipaddress/${EXTEN} |
| 01:03.56 | [hC] | ie, without brackets? |
| 01:04.14 | newmedian | Has anyone created their own AUP / terms of service, regarding what is not acceptable terminating a VOIP on a PSTN? e.g. sequential dialing, wardialing, telemarketing, etc. |
| 01:04.33 | |Vulture| | Anyone here use g729 |
| 01:04.45 | *** join/#asterisk Inv_arp (junya@adsl-3-244-124.mia.bellsouth.net) |
| 01:04.53 | ariel_ | |Vulture|, one of my customers does |
| 01:06.26 | ariel_ | Inv_arp, welcome just another person from my home town is on the line.... |
| 01:06.39 | Inv_arp | ariel_: heh whats happening man |
| 01:06.51 | |Vulture| | ariel_: its strange I bought 2 licenses today and when I use my 7960 with g729 and bridge a call it says "WARNING[15174]: codec_g729.c:180 g729tolin_framein: Out of G.729 Decoder Licenses!" even though I use ilbc/ulaw on bridged calls |
| 01:06.52 | [hC] | ariel_: hmm. i just tried adding a register line and i got refused.. |
| 01:07.13 | ariel_ | [hC], then you have something miss typed some place |
| 01:07.34 | [hC] | ariel_: when adding a register line into iax.conf, the syntax register => remoteuser:remotepass@remotehost |
| 01:07.45 | jontow | May 12 21:07:37 NOTICE[38977]: channel.c:1691 ast_set_write_format: Unable to find a path from gsm to speex |
| 01:07.45 | jontow | May 12 21:07:37 WARNING[38977]: file.c:779 ast_streamfile: Unable to open demo-echotest (format speex): No such file or directory |
| 01:07.48 | jontow | that sucks :o |
| 01:07.48 | jontow | hehe |
| 01:07.53 | [hC] | on remote host, you need a [remoteuser] context in iax.conf with username=remoteuser and secret=remoteuser, right? |
| 01:07.53 | ariel_ | |Vulture|, check to see where your lisc are being used. |
| 01:08.14 | *** join/#asterisk ClayReiche123 (~creiche@73-117.35-65.tampabay.res.rr.com) |
| 01:08.21 | ariel_ | jontow, did you install speex |
| 01:08.21 | jontow | vulture; 'show channels' ? |
| 01:08.24 | |Vulture| | ariel_: it only shows it being used for the sip client |
| 01:08.38 | jontow | hmm, no; i thought it was included and i found no clear directions on the wiki saying it weren't.. hmm |
| 01:08.42 | |Vulture| | but "show g729" displays "1/2 encoders/decoders of 2 licensed channels are currently in use" |
| 01:08.46 | ariel_ | check you lisc count |
| 01:08.49 | jontow | guess i'll have to take a looksee |
| 01:09.00 | *** join/#asterisk cmaj (~chris@65-37-6-42.nrp2.roc.ny.frontiernet.net) |
| 01:09.36 | [hC] | ariel_: was my description above correct? |
| 01:10.08 | ariel_ | jontow, show translation |
| 01:10.09 | *** part/#asterisk ClayReiche123 (~creiche@73-117.35-65.tampabay.res.rr.com) |
| 01:10.21 | |Vulture| | something seems messed up with g729 with * HEAD |
| 01:10.25 | nwhit | why would I get alot static on sip lines after the call is put on hold? |
| 01:10.31 | *** join/#asterisk tsp (~tyler@S01060080c825173c.vc.shawcable.net) |
| 01:10.38 | nwhit | and pick the call back up |
| 01:10.42 | tsp | can someone help mew tih alsa? I'm gonna clicks and pops when I use it with asterisk. |
| 01:10.49 | |Vulture| | "68.204.236.95 201 000d6557-04 00101/00102 g729 Rx: ACK" "0/0 encoders/decoders of 2 licensed channels are currently in use" |
| 01:11.12 | |Vulture| | I am in an echo test in g729 and it says no licenses are in use |
| 01:11.27 | ariel_ | [hC], looks right. |
| 01:11.31 | Nethab | it might only use them when decoding or encoding |
| 01:11.36 | Nethab | not passing it straight back |
| 01:11.58 | ariel_ | |Vulture|, do you have canreinvite=no or yes? |
| 01:12.04 | tsp | Nethab: hey, you use fwd or sipphone with asterisk right? how good is your sound coming through alsa on the console? |
| 01:12.10 | Nethab | yes |
| 01:12.18 | Nethab | i don't use the console |
| 01:12.24 | |Vulture| | ariel_: no |
| 01:12.27 | Nethab | i unloaded all unnessary modules |
| 01:12.38 | |Vulture| | ariel_: canreinvite=no |
| 01:13.11 | ariel_ | |Vulture|, brb diaper change argh |
| 01:13.14 | tsp | it's going click pop click click |
| 01:13.17 | Nethab | sound is fine with fwd, but sipphone sounds funny if i disable the jitter buffer |
| 01:13.25 | tsp | but I"m still hearing the sound |
| 01:13.36 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 01:13.38 | tsp | what jitter buffer? where would that be defined? |
| 01:13.56 | Nethab | it's in iax.conf |
| 01:14.20 | tsp | ok |
| 01:14.31 | |Vulture| | ariel_: np thanx for the help |
| 01:14.36 | tsp | I think its a alsa issue - opensound.com works better |
| 01:17.02 | tsp | is there a list of config file parameters? |
| 01:17.05 | tsp | e.g. for alsa.conf |
| 01:17.23 | Nethab | don't know |
| 01:18.39 | tsp | how do I see if the jitter buffer is enabled? |
| 01:19.19 | jontow | grep ^jitterbuffer /etc/asterisk/iax.conf |
| 01:19.21 | jontow | ;) |
| 01:19.34 | *** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 01:21.32 | *** join/#asterisk Mike (~mike@201.138.165.115) |
| 01:21.41 | Mike | someone knows if this message means im ok Channel 83: Individual Clear channel (Default) (Slaves: 83)? |
| 01:22.27 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 01:23.16 | [hC] | shido6: poke |
| 01:24.57 | ariel_ | |Vulture|, back. Can't belive so much stuff can come out of a little girl. |
| 01:26.12 | ariel_ | |Vulture|, have you tested with diallow=all and only allow=g729 for that device? |
| 01:26.57 | tsp | why is alsa so bad? |
| 01:27.21 | tsp | asterisk is doing all this static distortion stuff |
| 01:28.40 | Nethab | Attention everyone, i just wanted to make sure everyone has at least heard the name Asterlink, that is all |
| 01:28.59 | *** join/#asterisk TheEmperor (~user@203.114.48.47) |
| 01:29.04 | tsp | can anyone help wtih alsa? |
| 01:29.55 | ariel_ | tsp, sorry I have never used alsa. |
| 01:33.59 | *** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 01:35.18 | Sedorox | Anyone use a Broadmedia phone? |
| 01:35.18 | Sedorox | or have used? |
| 01:37.40 | newbien | tsp: did all the usual stuff like adjust volume, turn off speakers to prevent echo, and feedback? |
| 01:43.11 | *** join/#asterisk nn (~mikael@ip-wv-68-119-129-065.charterwv.net) |
| 01:43.15 | TheEmperor | can someone tell me if this is correct? : http://pastebin.ca/11639 |
| 01:44.12 | *** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 01:47.22 | *** join/#asterisk nwhit (~nwhit@64.50.35.78) |
| 01:48.16 | |Vulture| | ariel_: thats how i have it, and it connects but for some reason it just eats up licenses |
| 01:48.18 | nwhit | another crazy question.... why would I get horrible audio from sip phone to sip phone? I have tried several different codecs with no avail |
| 01:48.48 | |Vulture| | ariel_: it doesn't like when I bridge calls... just warns me but still works... very odd |
| 01:50.12 | |Vulture| | I am gunna go to STABLE and see if it works |
| 01:50.22 | nwhit | the bad audio is on the receiver of the phone that was called |
| 01:50.24 | nn | i need to set up my asterisk tonight |
| 01:50.43 | ariel_ | |Vulture|, we found that with only 2 we had trouble in transfing and other things like parking it took 3 lisc at many times. |
| 01:51.09 | TheEmperor | something weird is happening, when i check voicemail using sip it is clear. when i use iax2 it is all choppy, any ideas? |
| 01:51.22 | ariel_ | But for some reason once you get the 2nd phone up it only took one more. So one was 3 lisc and 2 was 4 then 3 phones was 5. |
| 01:51.38 | ariel_ | as the number were added we actuall got to an even count. |
| 01:52.05 | Mavvie | I'm not really sure what to do with that one |
| 01:52.13 | nwhit | Maveric, i get the same thing |
| 01:52.29 | nwhit | Maveric, but everything works well for me |
| 01:52.43 | nn | anyone in vegas? :) |
| 01:52.53 | nwhit | nn, i live in vegas |
| 01:54.46 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 01:54.55 | nwhit | nn, why? |
| 01:55.20 | |Vulture| | ariel_: ah so there is like a license overhead |
| 01:55.52 | nn | nwhit: just curious what the job market is like ou tthere atm |
| 01:55.56 | nn | debating moving back |
| 01:57.17 | ariel_ | |Vulture|, yes if you have zap channels. when we did a system with just sip and iax it seem to not have that problem. But I only use stable not head. |
| 01:57.48 | |Vulture| | ariel_: the funny thing is that it works... just makes warnings lol |
| 01:58.21 | ariel_ | yes inbetween there was a process that did not get the lisc. |
| 01:58.54 | ariel_ | in head you can via the dial plan pick your codec. Which is nice. But it's not enough for me to move to head. |
| 01:59.56 | Sato1 | anyone having problems with FWD to receive calls in an asterisk? |
| 02:00.32 | Sato1 | i can recibe calls from "call me" option, but when i use another account, fwd hangup before my * even answer |
| 02:01.03 | ariel_ | Sato1, are you using it via sip or iax2 |
| 02:01.22 | |Vulture| | ariel_: didn't know you could do that in the DP |
| 02:01.35 | |Vulture| | ariel_: do you know what the command is? |
| 02:01.36 | Sato1 | iax2 |
| 02:01.40 | ariel_ | |Vulture|, you can in head |
| 02:01.57 | HeppyCat | good evening\ |
| 02:02.00 | Sato1 | i use fwd -> asterisk at work -> asterisk at home |
| 02:02.08 | ariel_ | Sato1, what is your number? |
| 02:02.17 | *** join/#asterisk loud (~ariel@gw.honeypot.no) |
| 02:02.21 | Sato1 | 486533 |
| 02:02.31 | ariel_ | it's different number at home from work? |
| 02:02.46 | |Vulture| | ariel_: but do you know what the command is?? |
| 02:02.50 | |Vulture| | SetCodec()? |
| 02:03.10 | ariel_ | Sato1, I got your voicemail on your asterisk box. |
| 02:03.21 | Sato1 | got another number at home too, but i want to redirect from work to home too when nobody answers at work |
| 02:03.37 | ariel_ | |Vulture|, was just reading about it last week. I don't remember but I think it's in the wiki. |
| 02:04.04 | Sato1 | ariel_, that voicemail is not the asterisk box, its the fwd mailbox |
| 02:04.06 | ariel_ | Sato1, make an iax2 account between the boxes and forward them via your iax. |
| 02:04.10 | Sato1 | err. voicemail |
| 02:04.19 | TheEmperor | dial,zap/g1/$ |
| 02:04.23 | TheEmperor | opps |
| 02:04.31 | ariel_ | I got a menu and a voice that was not fwd system. |
| 02:04.40 | Sato1 | ariel, it already made |
| 02:04.54 | ariel_ | what do you get on the cli |
| 02:04.59 | TheEmperor | dial,Zap/g1/${EXTEN:1} : where do i specify g1? in zaptel.conf? |
| 02:05.08 | Sato1 | hold on, i will paste in a page |
| 02:05.37 | ariel_ | I get we are unable to answer your call now please leave your message. |
| 02:06.46 | *** join/#asterisk Defraz (~t0tal@65.103.222.4) |
| 02:07.02 | Defraz | I was curious if someone might point me in the right direction to cut down on echo? |
| 02:07.18 | Defraz | I get it from sip to sip from sip to pstn and pstn to sip of course. |
| 02:07.39 | ariel_ | Defraz, you have to start at echotraining=800 then work from there look up the settings on the wiki. |
| 02:07.39 | Defraz | I just don't know what to tweak anymore, does anyone have any suggestions? |
| 02:08.36 | ariel_ | how is your network? Do you have switch or hubs? |
| 02:08.55 | Defraz | switches and I even get it on my own controled network, bandwidth is plentiful. |
| 02:09.15 | *** join/#asterisk iq|laptop (~iq@65-103-164-141.omah.qwest.net) |
| 02:09.16 | Defraz | 10/100s and I am running and addiquit machine |
| 02:09.24 | ariel_ | Defraz, what is the server? |
| 02:09.27 | Qwell | addiquit? heh |
| 02:09.40 | ariel_ | what is an addiquit machine? |
| 02:09.47 | Sato1 | ariel_: http://cweb.wizardteam.com/fwd.htm |
| 02:09.57 | Qwell | adequate :) |
| 02:10.07 | Qwell | way off... |
| 02:10.27 | |Vulture| | ariel_: okay now I have 4 lics... just have to wait for them to be activated so I can try this again |
| 02:11.06 | Defraz | Intel(R) Pentium(R) 4 CPU 2.80GHz |
| 02:11.15 | ChulJin | well, nethab, that's one way to advertise |
| 02:11.26 | Defraz | thanks been wanting to know how to spell that. |
| 02:11.32 | Defraz | 2 gig ram |
| 02:11.44 | Defraz | figured for 20 extentions should be fine |
| 02:11.46 | *** join/#asterisk tessier (~treed@222.253.83.123) |
| 02:11.49 | Defraz | and a pri coming in. |
| 02:12.05 | Defraz | I am using the linksys PAP2 atas |
| 02:12.09 | Nethab | I'm not associated with them, but someone who's been in here for as long as tzanger has doesn't know about it... it's just sad |
| 02:12.14 | Defraz | everything works but i get echo. |
| 02:12.46 | ChulJin | nethab: it gets worse...I only just yesterday discovered they do term/orig...I thought they just did consulting. |
| 02:12.52 | Defraz | seems like my memory usage goes up in the 70s and 80s |
| 02:12.53 | ChulJin | and bkw apparently works for them. |
| 02:12.56 | ChulJin | that's all I knew. |
| 02:13.19 | nitram | weird thing... if i use the manager api to originate a call and that originated call is dialed through another * box through iax2 the extension dialed on the 2. * box looks like "<extension_dialed> <originating_extension> " while the original string was only "<extension_dialed>" |
| 02:13.20 | Nethab | that's more than tzanger knew, and he's been in here for 2 years |
| 02:13.28 | loud | Any news about broadvoice, when will they put their shit together ? |
| 02:13.38 | nitram | how come? |
| 02:13.53 | ChulJin | loud: ask ManxPower, he loves to discuss BroadVoice... |
| 02:14.06 | nitram | when i dial that same extension from a local phone everything is ok |
| 02:14.07 | *** join/#asterisk kimo_sabe (nick@zappa.azrackspace.net) |
| 02:14.12 | loud | will do |
| 02:14.54 | nitram | so the call inserted by the manager interface seems to be somewhat different than the one originated by a phone |
| 02:15.08 | ChulJin | loud: don't talk to ManxPower about broadvoice... |
| 02:15.16 | ChulJin | I was giving him a hard time... |
| 02:15.42 | loud | yeah, he owns a vo ip company right, i suppose he hates them |
| 02:15.48 | loud | but he might know! |
| 02:16.06 | loud | see, they have been down for like a week, thats very strange. |
| 02:16.40 | Nethab | did you see broadvoices letter |
| 02:16.58 | Qwell | "Dear Asterisk users. You suck. Signed - Broadvoice."? |
| 02:16.59 | nitram | damn... why is the callerid inserted there? |
| 02:17.16 | Nethab | they made a major upgrade, and then one of the carriers closed their account |
| 02:17.28 | Nethab | and charged the 13x overage charges |
| 02:17.49 | Qwell | excuses |
| 02:17.54 | Nethab | http://www.broadvoice.com/president_msg.html |
| 02:18.46 | [hC] | huh.. is there some problem with iax dialing in the last couple months of cvs head? for some reason i found that once i switched one of my iax contexts to use plaintext for auth, it was trying another context's password. when i removed that context, it stopped trying to send the password altogether. |
| 02:19.52 | loud | Nethab, thank you. |
| 02:20.01 | Nethab | for? |
| 02:20.21 | loud | didnt know that, the letter. |
| 02:20.46 | Nethab | yeah a bunch of us called in about downtime, and they sent us the letter |
| 02:21.47 | loud | ill re route to voipjet then. |
| 02:23.49 | *** join/#asterisk Inv_arp (junya@adsl-3-247-188.mia.bellsouth.net) |
| 02:25.20 | nitram | wtf is the callerid appended |
| 02:25.23 | ChulJin | 'depstein@broadvoice.com' is sooooo getting signed up for dozens of porn spam newsletters. |
| 02:27.13 | loud | im sure they must block those emails at a router level. |
| 02:28.27 | tsp | Whenever I dial an invalid fwd number - e.g. 123, the fast busy signal has a lot of distortion in it |
| 02:28.32 | tsp | and so does other sound |
| 02:28.35 | tsp | anyone know what's up? |
| 02:28.51 | [hC] | hm. okay maybe a better question. in an iax trunk, if you specify a plaintext password in secret= on both sides, and dont explicitly put auth=plaintext on both sides iax.conf, will it fail? |
| 02:29.25 | Nethab | the congestion should be coming from your local system not from fwd |
| 02:33.05 | *** join/#asterisk tessier (~treed@203.210.212.154) |
| 02:33.24 | ManxPower | But since the congestion tone IS coming from FWD there's nothing you can do about it. |
| 02:33.58 | ManxPower | I'm assuming you are using IAX to connect to FWD, of course. |
| 02:40.07 | meppl | gute nacht |
| 02:40.13 | Nugget | N8 |
| 02:40.13 | *** part/#asterisk Defraz (~t0tal@65.103.222.4) |
| 02:40.18 | meppl | gute nacht nugget |
| 02:41.47 | Sato1 | ManxPower, you are talking about my fwd problem? |
| 02:43.31 | *** part/#asterisk Gand_DJ (fabsced@ptr-207-54-104-24.ptr.terago.ca) |
| 02:43.39 | *** join/#asterisk file (~file@mctn1-3494.nb.aliant.net) |
| 02:45.59 | ManxPower | I simply spout words of wisdom. It's left as an exersize to the reader to determine of they apply or not. |
| 02:46.29 | AgiNamu | ChulJin, why |
| 02:46.43 | Sato1 | well, that apply to my case too, hehehe |
| 02:46.58 | AgiNamu | where's this open letter? |
| 02:47.12 | ChulJin | Agi: I was joking, of course. I have no problem with BV, because I sensibly avoid them. |
| 02:47.26 | ChulJin | ~broadvoice |
| 02:47.27 | jbot | extra, extra, read all about it, broadvoice is Follow the config instructions at http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup |
| 02:47.40 | [hC] | for call quality, which is better, g726, speex, or gsm |
| 02:47.50 | AgiNamu | g726 prolly |
| 02:48.13 | *** join/#asterisk grolloj (~grolloj@dsl254-116-106.nyc1.dsl.speakeasy.net) |
| 02:48.18 | [hC] | freeworldtel seems to be using gsm for some reason, even though i am preffing g726 |
| 02:48.51 | ChulJin | jbot: no, broadvoice is Follow the config instructions at http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup but also beware: http://www.broadvoice.com/president_msg.html |
| 02:48.53 | jbot | ChulJin: okay |
| 02:48.57 | ChulJin | ~broadvoice |
| 02:48.58 | jbot | from memory, broadvoice is Follow the config instructions at http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup but also beware: http://www.broadvoice.com/president_msg.html |
| 02:50.47 | ManxPower | When we were using G726 someone asked the MIS manager what phone system he was using because the voice quality was so good. |
| 02:51.03 | ManxPower | Of course, it was not any better than ulaw/alaw. It could not be. |
| 02:51.18 | Juggie | yeah |
| 02:51.28 | Juggie | funny people who think g729 is better then ulaw |
| 02:51.33 | ManxPower | But it is an indication of how close G726 sounds to ulaw/alaw |
| 02:51.36 | ChulJin | wow, speex is so expensive |
| 02:51.37 | Qwell | ManxPower: I've found that sometimes when you have a low quality MP3 (96kbps?), some people find that it sounds good |
| 02:51.40 | ChulJin | er 'expensive' |
| 02:51.44 | [hC] | Yeah, i notice that when using voipjet with ulaw, occasionally i get choppiness, and sometimes i echo back to myself a touch.. |
| 02:51.45 | Juggie | too bad everything is sampled into ulaw first |
| 02:51.51 | ChulJin | juggie: g729 is better because it is not free. |
| 02:51.55 | AgiNamu | So... in other words... don't buy your wholesale termination from retail vendors? :P |
| 02:52.01 | [hC] | using freeworldtel (link2voip) they pref 726, speec, and gsm, and of course the call fell back to gsm |
| 02:52.42 | [hC] | aha it seems to be reading it bottom to top, for pref |
| 02:53.06 | [hC] | is that normal in iax.conf? i thought it read top to bottom |
| 02:53.06 | Qwell | [hC]: its not "in order" I don't believe |
| 02:53.16 | Juggie | wow |
| 02:53.16 | Qwell | Its a hard coded order afaik |
| 02:53.23 | Juggie | i actually found a sip phone that correctally follows the spec |
| 02:53.27 | [hC] | it must be on their side that the order is chosen. |
| 02:53.30 | Juggie | and doesnt try and be lame and do fucked up shit |
| 02:53.34 | Juggie | aka xten/firefly |
| 02:53.39 | [hC] | Juggie: which? |
| 02:53.43 | Juggie | sjphone |
| 02:53.55 | [hC] | ah. softphone. |
| 02:53.56 | Juggie | xten tries to help nat, but it just screws me over |
| 02:54.08 | Juggie | i want it to come in on the rtp port asterisk tells it to via sdp |
| 02:54.11 | Juggie | but it does its own thing |
| 02:54.17 | Juggie | and picks other random ports |
| 02:54.47 | AgiNamu | What's a TN? |
| 02:54.55 | [hC] | hrm yeah cause i had a disallow=all, allow=g726, allow=speex, allow=gsm.. it was picking gsm. when i commented the gsm line out, it picked 726. |
| 02:55.37 | [hC] | so i guess preference cannot be decided on the end users part. |
| 02:56.02 | Qwell | [hC]: read what I said... |
| 02:56.14 | [hC] | Qwell: okay. |
| 02:56.41 | [hC] | I was under the impression that a disallow=all, and allow lines would imply preference order. I guess its just to specifically set codecs that can be used then. |
| 02:56.42 | [hC] | thanks |
| 02:56.54 | Qwell | There was a patch or something for that though, wasn't there? |
| 02:57.00 | [hC] | Ya got me. |
| 02:57.19 | [hC] | Isnt a register => line only necessary if the peer has a dynamic ip? |
| 02:57.26 | [hC] | in iax.conf |
| 02:57.30 | WilliamK | anyone know if there is a wave file for call blocking? |
| 02:57.35 | WilliamK | err wav |
| 02:58.07 | [hC] | i use this: |
| 02:58.08 | [hC] | exten => s,3,Playback(privacy-this-number-is) |
| 02:58.08 | [hC] | exten => s,4,Playback(privacy-blocked) |
| 02:58.42 | WilliamK | hmm, I don't have those wav's |
| 02:58.51 | [hC] | they're from the sound addons from cvs |
| 02:59.31 | WilliamK | ah |
| 03:00.31 | WilliamK | um ok, how do I see a list of the tree for cvs? |
| 03:03.26 | syle | are there any examples to show how to call in your asterisk at home with your cell phone , then dial a long distance number? |
| 03:03.45 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 03:05.10 | Sato1 | i think i found the problem! |
| 03:05.14 | Sato1 | well i did |
| 03:06.20 | Sato1 | FWD is the problem as ManxPower said |
| 03:06.42 | Sato1 | i just setup other account, and it is working fine, dont know whats wrong with my old account |
| 03:08.26 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 03:12.17 | TheEmperor | can anyone help me? when i use iax2 to check my mail, the sound is all choppy but using sip no problem.. |
| 03:16.16 | Nethab | are you jittery? |
| 03:16.45 | TheEmperor | i turned jitterbuffer off in iax.conf |
| 03:17.15 | TheEmperor | but i might have done something wrong in extensions.conf, i am not sure.. |
| 03:17.18 | TheEmperor | http://pastebin.ca/11639 |
| 03:19.21 | Juggie | TheEmperor, how did your new extensions.conf work out for you |
| 03:19.35 | TheEmperor | Juggie: sip is working good, but problems with iax |
| 03:19.35 | blitzrage | if the sound is choppy, its not extensions.conf |
| 03:19.51 | TheEmperor | blitzrage: but when i use sip it's fine.. |
| 03:19.55 | blitzrage | exactly |
| 03:20.02 | blitzrage | then its not extensions.conf |
| 03:20.12 | blitzrage | turn jitterbuffer back on and turn trunking off |
| 03:20.22 | TheEmperor | juggie: still can't dial out though...even after defining group= 1 |
| 03:20.22 | blitzrage | do one at a time and see if its any better |
| 03:20.36 | TheEmperor | blitzrage: in zaptel.conf? |
| 03:20.37 | blitzrage | yo file |
| 03:20.42 | file | how are you? |
| 03:20.44 | blitzrage | TheEmperor: you said iax - so iax.conf |
| 03:20.51 | TheEmperor | ok |
| 03:21.00 | blitzrage | file: not too bad, just about to go do some more PHP / PostgreSQL programming |
| 03:21.15 | blitzrage | TheEmperor: zaptel.conf configures zaptel hardware and has nothing to do with IAX. |
| 03:21.26 | TheEmperor | ok |
| 03:21.45 | blitzrage | file: it should be a lot easier to answer these questions whenever new book comes out :) |
| 03:21.53 | blitzrage | a new book* |
| 03:22.02 | TheEmperor | blitzrage: when i use iax with musiconhold it's fine, it's just voicemail.. |
| 03:22.26 | TheEmperor | Juggie: can you have a quick look at my iax extensions? http://pastebin.ca/11639 |
| 03:22.29 | blitzrage | TheEmperor: not too sure - thats screwed up :) |
| 03:22.30 | TheEmperor | :) |
| 03:22.43 | file | blitzrage: are you still puttering away on things? |
| 03:22.44 | blitzrage | never had that problem before |
| 03:22.52 | TheEmperor | blitzrage: i even tried to update using cvs and all, but still no good :( |
| 03:22.53 | blitzrage | file: yah, puttering is a good word :) |
| 03:22.57 | file | blitzrage: hehe |
| 03:23.16 | blitzrage | file: I believe how long it took me to write like 50 lines of code last night |
| 03:23.35 | blitzrage | file: just couldn't figure out what the hell the problem was - turns out I was putting a <form> inside another <form> that I didn't notice |
| 03:23.43 | file | pure sillyness |
| 03:23.46 | Weezey | can anyone tell me how to make * show me why a module won't load? |
| 03:24.18 | *** join/#asterisk implicit (~implicit@dhcp-114-131.mobile.uci.edu) |
| 03:24.24 | blitzrage | /usr/sbin/asterisk -vvvvvvvvvvv | tee /tmp/debug.log |
| 03:24.25 | JerJer | Weezey: asterisk -vvvgc |
| 03:24.36 | Weezey | thanks blitzrage |
| 03:24.45 | blitzrage | listen to JerJer too, he knows more than me :) |
| 03:24.51 | implicit | lol |
| 03:25.04 | blitzrage | implicit: lol, whats up? :) |
| 03:25.09 | implicit | whats up man |
| 03:25.15 | implicit | just been coding a lot |
| 03:25.16 | implicit | :) |
| 03:25.24 | implicit | SER modules ;) |
| 03:25.33 | blitzrage | not too much - just sitting back down at the computer after swimming for an hour then resting on the couch |
| 03:25.42 | Weezey | JerJer: actually maybe you can help me, chan_h323.so isn't loading I've compiled using correct versions of pwlib and openh323, latest CVS. |
| 03:25.48 | blitzrage | implicit: oh fun, I'm about to continue working on an Asterisk GUI |
| 03:25.54 | TheEmperor | can anyone help, i am still having problems dialling out on Zap channels even when I have already defined that in zapata.conf as group=1 |
| 03:26.01 | implicit | blitzrage, yeah it's really good stuff |
| 03:26.05 | blitzrage | implicit: meh :) |
| 03:26.13 | implicit | :) u don't like SER????? |
| 03:26.18 | blitzrage | implicit: its gotta be done :) |
| 03:26.25 | file | yay SER |
| 03:26.25 | TheEmperor | in extensions file it's dial,Zap/g1/${EXTEN} |
| 03:26.29 | blitzrage | implicit: oh, we're talking about SER now? :) |
| 03:26.33 | implicit | i thought we were |
| 03:26.35 | implicit | file, right on |
| 03:26.36 | blitzrage | lol |
| 03:26.44 | blitzrage | I don't know what i'm talking about anymore |
| 03:26.46 | implicit | file, thanks for those ideas btw |
| 03:26.52 | file | yw |
| 03:27.04 | *** join/#asterisk K9DI_BSD_WrkStn (~k9bsd@207-246-185-168.EastVillage.ResNet.wiu.edu) |
| 03:27.13 | blitzrage | implicit: I liked SER when I used it once - just haven't a need for it right now |
| 03:27.25 | blitzrage | implicit: what are you using it for? |
| 03:27.38 | blitzrage | I'm always curious what people use it for :) |
| 03:27.55 | _DAW | TheEmperor - Errors? |
| 03:28.07 | implicit | blitzrage, for everything |
| 03:28.14 | implicit | blitzrage, tell me what i am NOT using it for |
| 03:28.15 | implicit | lol |
| 03:28.19 | blitzrage | haha |
| 03:28.24 | implicit | sems for error announcements and stuff |
| 03:28.36 | implicit | but all routing, billing, everything |
| 03:28.41 | blitzrage | hrmmmm, interesting |
| 03:28.49 | file | it can do it all, you just have to write it. |
| 03:28.54 | implicit | file, yep |
| 03:29.00 | blitzrage | I already have too much stuff to learn and document in Asterisk - I don't need another project :) |
| 03:29.02 | implicit | not too hard to write anyway |
| 03:29.04 | implicit | *either |
| 03:29.05 | implicit | i meant |
| 03:29.17 | implicit | IVRs are easy/nice in sems too |
| 03:29.19 | TheEmperor | _DAW: Errors when dialling out you mean? |
| 03:29.28 | file | the asterlink build I have going is cute :) |
| 03:29.32 | _DAW | yes |
| 03:29.59 | TheEmperor | _DAW: It just hangs when I dial out.. |
| 03:30.47 | nwhit | another crazy question.... why would I get horrible audio from sip phone to sip phone on a local 100mbit lan? I have tried several different codecs with no avail |
| 03:31.14 | TheEmperor | _DAW: this is what I get |
| 03:31.16 | TheEmperor | <PROTECTED> |
| 03:31.16 | TheEmperor | <PROTECTED> |
| 03:31.16 | TheEmperor | <PROTECTED> |
| 03:31.16 | TheEmperor | <PROTECTED> |
| 03:31.33 | TheEmperor | the thing is, the phone line is plugged into zap 4... |
| 03:31.35 | Juggie | yay |
| 03:31.56 | Juggie | it wasnt easy but i made sipphone<-nat->internet<-nat/dmz->asterisk work |
| 03:33.09 | Juggie | TheEmperor, your extensions look fine.. |
| 03:33.19 | TheEmperor | Thanks Juggie :) |
| 03:33.32 | Juggie | you never got groups working? |
| 03:33.38 | TheEmperor | no... |
| 03:33.42 | TheEmperor | still doesn't work... |
| 03:33.48 | Juggie | pastebin your zapata.conf |
| 03:33.54 | TheEmperor | ok |
| 03:33.59 | _DAW | yes please pastebin |
| 03:34.17 | *** join/#asterisk docelmo (~me@116-39.202-68.tampabay.res.rr.com) |
| 03:34.19 | WilliamK | nwhit, check your duplex settings for the nics and make sure they're 100full and not mismatched if you're using a switch |
| 03:34.31 | blitzrage | WilliamK: oooo, good thinking :) |
| 03:34.33 | docelmo | question.. Does the DIALSTATUS variable work with oh323? |
| 03:34.50 | blitzrage | docelmo: its set with Dial() |
| 03:34.54 | Juggie | dialstatus is dependant on app_dial not the protocol |
| 03:35.03 | blitzrage | afaik |
| 03:35.11 | TheEmperor | http://pastebin.ca/11644 |
| 03:35.22 | docelmo | Thats what I thought.. Cause.. its not working.. When was it implimented? |
| 03:35.27 | docelmo | what version? |
| 03:35.30 | WilliamK | blitz, I recently had to write it into the sysinit, because the catalyst 5500 was doing one thing and the linux boxes were doing another |
| 03:35.37 | nwhit | WilliamK, on the asterisk box? |
| 03:35.46 | nwhit | WilliamK, or the phones |
| 03:36.00 | WilliamK | nwhit, start on the asterisk box |
| 03:36.33 | WilliamK | if it's half duplex, it could be "slightly impairing" not usually enough to notice though |
| 03:36.51 | Juggie | TheEmperor, why do you have context more then once.... you only need it once |
| 03:36.56 | Juggie | local-in is all you need |
| 03:37.01 | WilliamK | however if you're switch is talking half duplex, and your box is full, it'll act like a 56k modem |
| 03:37.07 | WilliamK | IF it works at all |
| 03:37.14 | TheEmperor | Juggie: no need for group =1? |
| 03:37.36 | nwhit | WilliamK, where do I find that... sorry never have looked for it before |
| 03:37.40 | _DAW | No need to two contexts |
| 03:37.44 | WilliamK | what version of linux? |
| 03:37.51 | nwhit | fc3 |
| 03:37.53 | TheEmperor | i see |
| 03:38.01 | WilliamK | mii-tool or ethtool |
| 03:38.02 | _DAW | group one should be for your local-out |
| 03:38.12 | Juggie | TheEmperor, no, no need for context=local-out |
| 03:38.34 | Juggie | you dont have analog phones |
| 03:38.40 | TheEmperor | do i still need group = 1 and so on? |
| 03:38.49 | Juggie | yes |
| 03:38.52 | TheEmperor | ok |
| 03:38.53 | nwhit | WilliamK, looks like i am 100baseT full duplex |
| 03:39.08 | nwhit | so you think i should drop it to half? |
| 03:39.13 | Juggie | no |
| 03:39.15 | Juggie | why would you do that |
| 03:39.17 | WilliamK | definately not |
| 03:39.34 | docelmo | is there something wrong with the cvs? |
| 03:39.47 | Juggie | TheEmperor, i dont know why that group woudnt take, anyone know? |
| 03:39.49 | WilliamK | half duplex = collissions, interface resets, etc.. = BAD |
| 03:40.00 | nwhit | WilliamK, that's what I though |
| 03:40.08 | WilliamK | all bad things for voip/data packets |
| 03:40.15 | nwhit | WilliamK, I have actually tried a different switch also |
| 03:40.23 | TheEmperor | Juggie: does asterisk look for lines in zap 1 and then if it's unavailable go to zap 2,3 and so on? |
| 03:40.40 | Juggie | TheEmperor, if you did Zap/g1 and your 4 channels were in that group |
| 03:40.41 | Juggie | then yes |
| 03:40.51 | JerJer | um no |
| 03:40.52 | Juggie | when you do a dial on Zap/g1 |
| 03:40.54 | Juggie | what happens |
| 03:40.54 | WilliamK | nwhit, unmanaged or managed switch? |
| 03:41.08 | TheEmperor | well, there is no phone line plugged in there, so it just hangs |
| 03:41.10 | nwhit | WilliamK, i just did a flood ping to the phone and got 100% response with .5ms response avg |
| 03:41.13 | Juggie | ah |
| 03:41.13 | Juggie | well |
| 03:41.14 | TheEmperor | the line is plugged into line 4 |
| 03:41.15 | nwhit | WilliamK, unmanaged |
| 03:41.21 | Juggie | remove that line from the group :) or just dial on line 4 |
| 03:41.22 | Juggie | hah |
| 03:41.44 | TheEmperor | i guess the question is, if zap 1 was busy, will the call go out on zap 2 ? |
| 03:41.46 | nwhit | WilliamK, the calls sound great going out through a zap channel it is just sip-sip in the office |
| 03:41.48 | Juggie | yes |
| 03:41.51 | TheEmperor | i see |
| 03:41.58 | JerJer | busy is different than simply no phone line |
| 03:41.59 | TheEmperor | so i should plug that line into zap 1 instead of 4 :) |
| 03:41.59 | Juggie | if zap chan1 was offhook |
| 03:42.06 | Juggie | it would move to chan2 and so on |
| 03:42.08 | TheEmperor | i see isee |
| 03:42.10 | TheEmperor | dumb me! |
| 03:42.11 | WilliamK | nwhit, check your packet payload sizes and make sure they're the same |
| 03:42.28 | Juggie | JerJer, i didnt realize he only had a line in the last port |
| 03:42.28 | WilliamK | worse comes to worse, try g711u until you get it figured out at least |
| 03:42.42 | nwhit | WilliamK, 20ms both |
| 03:43.02 | nwhit | WilliamK, that is the weird part.... i've tried g711u, gsm, g726 ... all the same |
| 03:43.13 | nwhit | WilliamK, i am thinking it might be my phones (Snom 360) |
| 03:43.27 | Juggie | nwhit, is canreinvite=yes ? |
| 03:43.33 | Juggie | so the rtp goes phone to phone |
| 03:43.37 | WilliamK | nwhit, got the newest firmware? |
| 03:43.40 | nwhit | Juggie, tried it both ways |
| 03:43.46 | nwhit | WilliamK, the latest stable |
| 03:43.53 | WilliamK | try an alpha |
| 03:44.01 | CoaxD | smokin' the ganj! |
| 03:44.04 | Juggie | if canreinvite=no |
| 03:44.06 | WilliamK | yeah, that word scares some people |
| 03:44.07 | Juggie | i'd blame asterisk |
| 03:44.14 | Juggie | and say you had an irq conflict or something |
| 03:44.16 | Juggie | causing problems |
| 03:44.22 | Juggie | but you tried it both ways |
| 03:44.44 | nwhit | Juggie, it works rather well throught the asterisk box because the sip->zap works great |
| 03:45.03 | nwhit | ok... i'm going to try to upgrade the phones... hold on a sec |
| 03:45.16 | WilliamK | which ver you trying? |
| 03:45.30 | CoaxD | Damn. in the last 2 hours, i've been doing nothing, and I earned $62.82 |
| 03:45.32 | TheEmperor | ah... |
| 03:45.34 | TheEmperor | now it works... |
| 03:45.36 | TheEmperor | :) |
| 03:45.49 | TheEmperor | i was thinking that asterisk could tell which port i plugged the line into! |
| 03:46.01 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 03:46.29 | nwhit | 3.60i |
| 03:46.55 | nwhit | i was using 3.60b |
| 03:47.09 | WilliamK | ah b is bad |
| 03:47.16 | WilliamK | least on the 190s |
| 03:47.16 | nwhit | really? |
| 03:47.26 | nwhit | WilliamK, like what? |
| 03:47.40 | WilliamK | my phone did all kinds of wacky things |
| 03:47.50 | nwhit | what are you using now? |
| 03:47.51 | WilliamK | ALOT of fixes in "i" though |
| 03:48.00 | WilliamK | SNOM 190, and Sipura ATA |
| 03:48.23 | nwhit | i really like the sipura atas |
| 03:48.36 | nwhit | i wonder what cisco is going to do to them after the buyout |
| 03:49.21 | WilliamK | prolly keep going along as normal I would assume |
| 03:49.26 | WilliamK | or integrate them into Linksys |
| 03:49.40 | WilliamK | since afterall they do have a large intercompany contract now |
| 03:50.01 | nwhit | yeah |
| 03:50.32 | WilliamK | by the way, the 3.60d was even so bad I downgraded to 3.56i |
| 03:50.37 | WilliamK | on the 190 |
| 03:50.42 | WilliamK | customers had a hard time hearing me |
| 03:51.17 | nwhit | hold on |
| 03:51.20 | WilliamK | err 3.56y not 3.56i |
| 03:53.58 | nwhit | i think it might be this particular phone |
| 03:54.48 | *** join/#asterisk james_ed (~james_ed@ip68-103-200-171.ks.ok.cox.net) |
| 03:55.01 | james_ed | hello all |
| 03:55.27 | WilliamK | nwhit, could always try and factory default it back |
| 03:55.50 | nwhit | WilliamK, that is what i am doing right now |
| 03:56.09 | nwhit | WilliamK, the phone was acting kinda weird... locking up sometimes on bootup, etc |
| 03:57.33 | Juggie | nwhit, have you tried with qualify on or off |
| 03:58.00 | *** join/#asterisk tiko_007 (~tiko_007@218.108.174.21) |
| 04:01.14 | nwhit | Juggie, nope |
| 04:02.06 | james_ed | question...anyone familiar with cisco 9710...is there a sip load for it? i don't see one on cisco's site |
| 04:03.11 | Juggie | i had a phone that crashed without qualify=yes |
| 04:04.16 | nwhit | hahaha... someone gave me a crossover cable to use for this phone.... |
| 04:04.50 | Juggie | shoudnt even work |
| 04:04.59 | JerJer | 7910 = SCCP only |
| 04:05.05 | Juggie | unless your switch does some funky auto sensing shit |
| 04:05.13 | WilliamK | actually I could see where it might though...MDI-X |
| 04:05.15 | WilliamK | =) |
| 04:05.16 | Juggie | how is SCCP support in * havnt tried it |
| 04:05.28 | JerJer | ymmv |
| 04:06.09 | nwhit | it sorta worked |
| 04:06.22 | Juggie | i use mitel 5220's now |
| 04:06.24 | Juggie | solid phone |
| 04:06.33 | Juggie | only complaint is lack of a remote directory support |
| 04:06.41 | JerJer | i haven't played with chan_skinny in ages |
| 04:07.05 | Juggie | there are too many chan_h323 drivers now |
| 04:07.21 | Juggie | something needs to be done to form a consolidated project |
| 04:07.39 | nwhit | i am really kicking myself right now |
| 04:07.57 | Juggie | y? |
| 04:08.00 | nwhit | i thought i should try another cable... but noooo... the cable couldn't cause this.... |
| 04:08.19 | nwhit | but behold... it is the cable... |
| 04:08.26 | nwhit | i am amazed it even worked too |
| 04:08.41 | Juggie | odd |
| 04:08.46 | sudhir492 | How big extensions.conf and sip.conf files will affect asterisk's performance badly? |
| 04:09.10 | Juggie | sudhir492, probally bigger then you need, but i would suggest using includes for readability purposes |
| 04:11.29 | sudhir492 | I have around 100 entries now. Can expect it to grow to about 3500 (whole alumni association) with every extension practically having its own forwarding number. Can asterisk handle that without any decrease in performance |
| 04:12.22 | Juggie | sudhir492, are you just giving people internal extensions and redirecting to an external phone number? |
| 04:12.59 | Juggie | or are you giving everyone sip phones etc |
| 04:13.43 | Juggie | <waits> |
| 04:13.46 | Juggie | i dont like waiting |
| 04:13.48 | sudhir492 | everyone has a SIP device too with internal number |
| 04:14.08 | Juggie | soft phone? or hard phone? |
| 04:14.24 | sudhir492 | but people like to forward on no-answer to their alternative numbers |
| 04:14.34 | Juggie | understood |
| 04:14.43 | sudhir492 | some people are quite excited that they can make 3 phones ring at the same time :-) |
| 04:14.55 | Juggie | indeed |
| 04:14.58 | ManxPower | Cisco's web site SAYS the 7910 supports SIP, however Cisco does not have SIP firmware for the 7910. |
| 04:14.58 | Juggie | regardless |
| 04:15.08 | Juggie | before you get too excited |
| 04:15.13 | Juggie | provide me with some details |
| 04:15.17 | ManxPower | This issue was one of the first "rude awakenings" I had in the VoIP world. |
| 04:15.29 | Juggie | everyone has a sip phone which is a soft phone? |
| 04:15.59 | Juggie | ManxPower, if your a cisco customer, call them up and ask for it |
| 04:16.02 | sudhir492 | some people have soft phones, but most of them will Grandstream or Sipura |
| 04:16.04 | ManxPower | sudhir492: with 3500 phones Asterisk may take a while to start, but it should work fine after than. |
| 04:16.27 | ManxPower | Juggie: Um, I have a full access support contract. There is NO SIP firmware for the 7910. It's a documentation bug on their web sire. |
| 04:16.28 | ManxPower | site. |
| 04:16.36 | Juggie | i see |
| 04:16.47 | Juggie | sudhir492, first thing i will recomend is to NOT write a huge dialplan |
| 04:16.55 | Juggie | eg, give the phones names |
| 04:17.04 | Juggie | and have to resolve extensions to names in dialplan |
| 04:17.06 | Juggie | that would be hell |
| 04:17.23 | sudhir492 | All extensions are numbers. |
| 04:17.26 | ManxPower | Juggie: The SIP user ID of each of our SIP phones is the MAC address. |
| 04:17.30 | sudhir492 | going to be numbers |
| 04:17.43 | Juggie | so ,the phone registeres as 1000 or something like that. |
| 04:18.06 | Juggie | manx, good setup, but i prefer using extensions as the user |
| 04:18.13 | Juggie | that way dialplan needs no extra work |
| 04:18.14 | sudhir492 | However, for most of the phones there are going to be forward on no-answer |
| 04:18.52 | Juggie | your dialplan should be small then |
| 04:19.02 | Juggie | and sip.conf large |
| 04:19.35 | ManxPower | Juggie: We used to do that. |
| 04:19.51 | ManxPower | Then we realized that we were limiting ourselves by thinking a phone was an extension |
| 04:19.52 | *** join/#asterisk hypa7ia (~leigh@69.158.182.171) |
| 04:20.04 | Juggie | manx, i found that doing it that way eg making the phone register as 4010 for example, then you can do something like |
| 04:20.17 | *** join/#asterisk mapu (~mapu@c-24-63-115-52.hsd1.ma.comcast.net) |
| 04:20.18 | Juggie | _ZXXX,1,Dial(SIP/${EXTEN}) |
| 04:20.24 | Juggie | and that covers all your phones :) |
| 04:20.38 | Juggie | if that dial fails, then the sip phone doesnt exist, or is not connected |
| 04:20.43 | ManxPower | Juggie: and then you are locked into all phones being handled exactly the same way in the dialplan. |
| 04:20.43 | WilliamK | but that's not always a good thing |
| 04:21.00 | Juggie | ManxPower, that suits most organizations though |
| 04:21.11 | Juggie | 90% of places want ring, voicemail |
| 04:21.11 | sudhir492 | For example, I expect to see about 3 lines for most of the phones: |
| 04:21.12 | sudhir492 | exten => 2000,1,Dial(SIP/${EXTEN}|25) |
| 04:21.12 | sudhir492 | exten => 2000,2,Dial(Zap/g1/17037980259&Zap/g1/7039730089|20) |
| 04:21.12 | sudhir492 | exten => 2000,3,voicemail(2000) |
| 04:21.25 | ManxPower | what is 1000 wants call forward on busy, and 1002 wants no voicemail, and 1007 wants three devices to ring? |
| 04:21.46 | Juggie | manx, thats still not an issue |
| 04:22.04 | Juggie | you could customize dialplan for them by hardcoding it, or writing some logic connected to a database |
| 04:22.05 | Juggie | which ever |
| 04:22.35 | ManxPower | and one user wants line 1 on their phone to be extension 1000, but line 2 to be the same as their boss and have it ring at the same time. |
| 04:22.51 | ManxPower | I just decided that thinking of phones as extensions was limiting our design. |
| 04:23.03 | Juggie | i think it depends on the application |
| 04:23.23 | Juggie | ManxPower, how do you share a line though |
| 04:23.30 | ManxPower | Juggie: I also figure that by the time we run into dialplan limitations we would be using a database. |
| 04:23.53 | ManxPower | Juggie: You don't. exten => 1000,1,Dial(SIP/MAC1&SIP/MAC2) |
| 04:23.59 | Juggie | thats what i thought |
| 04:24.04 | Juggie | i thought you might have had something better |
| 04:24.25 | ManxPower | rather than exten => 1000,Dial(SIP/1003&SIP/1000) |
| 04:24.52 | Juggie | yes... |
| 04:24.56 | Juggie | i agree.... |
| 04:25.07 | Juggie | when you want alot of flexiblity that would make things easier |
| 04:25.17 | ManxPower | Someone calls up with a problem and I say What is the MAC address of the phone? Rather than "what's your extension?" 8-) |
| 04:25.25 | Juggie | i would personally like to write a dialplan connected to mysql for controling your extension |
| 04:25.40 | ManxPower | The MAC address is, of course, printed on the bottom of every phone. |
| 04:25.48 | Juggie | yes.... |
| 04:26.04 | Juggie | ManxPower, you know something the voip world needs is a templating tftp server |
| 04:26.19 | ManxPower | Juggie: Something like the way Polycom does it? |
| 04:26.34 | *** join/#asterisk nolan-- (~chatzilla@216.199.159.79) |
| 04:26.34 | Juggie | what is their solution, i've not used any polycom sip phones |
| 04:27.06 | Juggie | i envisioned something like a tftp server connected to a database, i requested a certain file, and it just used a template + database record to build the file and send it to me |
| 04:27.24 | kimo_sabe | Juggie: I saw one of those somewhere.... I don't remember where though |
| 04:27.39 | ManxPower | Juggie: Polycom has systemwide default config files and a per-phone config file to override any system-wide defaults. |
| 04:27.48 | nwhit | ok... so it wasn't the cable after all.... I think it might be the phone, though |
| 04:27.51 | ManxPower | all downloaded via tftp |
| 04:28.09 | kimo_sabe | Zultys does that too |
| 04:28.34 | Juggie | manx, cisco does that |
| 04:28.37 | Juggie | and so does mitel |
| 04:28.44 | Juggie | but i want to remove those files all together |
| 04:28.49 | Juggie | one file, only |
| 04:28.58 | Juggie | and a database |
| 04:29.00 | kimo_sabe | it still would be nice to pull the display names and other configs from a RDBMS, or LDAP |
| 04:29.45 | Juggie | so imagin you have a database record with |
| 04:30.07 | Juggie | SIPSOMEMAC.cnf,thetemplate,value1,value2, and so on |
| 04:30.27 | Juggie | and the tftp sees the file your asking for, loads the template, fills in the missing values which are in the database and sends it along |
| 04:31.05 | JerJer | so write your own tftp server app |
| 04:31.19 | JerJer | mine is a very simple perl script |
| 04:31.22 | kimo_sabe | Juggie: tftp is a simple protocol, query your DB based on the MAC requesting |
| 04:32.17 | Juggie | yah i'm just a busy/lazy guy ;) |
| 04:35.06 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 04:35.06 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 04:35.17 | Deryl | FreeBSD Users - Disable hyperthreading: ftp://ftp.freebsd.org/pub/FreeBSD/CERT/advisories/FreeBSD-SA-05:09.htt.asc |
| 04:35.29 | Deryl | (for those that don't know already) |
| 04:36.23 | Sedorox | lol |
| 04:36.30 | Deryl | hehe, hey Sedorox |
| 04:36.35 | Sedorox | hola :p |
| 04:36.46 | Mavvie | Deryl: the funny thing is, it's an i386 attack which works on all machines with an i386 inside. |
| 04:36.47 | Deryl | least you can't say i don't make sure ;) |
| 04:36.57 | *** join/#asterisk ta[i]nted (~tainted@adsl-69-108-107-206.dsl.irvnca.pacbell.net) |
| 04:37.03 | Sedorox | yupo |
| 04:37.03 | Sedorox | :p |
| 04:37.36 | Deryl | Mavvie: umm, it's against specific CPUs |
| 04:37.43 | Deryl | Intel Pentium Extreme Edition, Pentium 4, Mobile Pentium 4, and Xeon processors |
| 04:37.49 | bkw_ | I seen the funniest shit at wal-mart tonight |
| 04:37.53 | bkw_ | "Home Drug Test" |
| 04:37.53 | Mavvie | okay, but it's not FreeBSD specific. |
| 04:38.05 | bkw_ | how fucking south eastern oklahoma do ya gotta get |
| 04:38.20 | Deryl | i never said it was. it IS however a freebsd alert |
| 04:38.23 | Juggie | bkw, have you ever heard of xlite being fucking stupid when it comes to rtp ports |
| 04:38.25 | Deryl | and many here run freebsd |
| 04:38.26 | Sedorox | hmmmmmm |
| 04:38.37 | Juggie | i spent hours trying to get phone-nat-internet-nat/dmz-asterisk working |
| 04:38.39 | Mavvie | Deryl: http://www.daemonology.net/hyperthreading-considered-harmful/ |
| 04:38.48 | Mavvie | only four groups have responded. |
| 04:38.51 | Juggie | only to discover xlite was ignoring the rtp port asterisk said (and my firewall rules for set for) |
| 04:38.52 | bkw_ | Juggie, nope |
| 04:38.56 | Mavvie | not a single linux distribution. |
| 04:39.00 | Deryl | Mavvie: and your point is.. ? |
| 04:39.02 | Juggie | it was sending to like port 3000ish |
| 04:39.28 | Mavvie | Deryl: still, that it is not FreeBSD specific. |
| 04:39.33 | Juggie | found the setting in xlite,it was "obey reverse udp mapping rules" |
| 04:39.36 | Juggie | turned that off |
| 04:39.38 | Deryl | Mavvie: and where did I say that it was? |
| 04:39.39 | Juggie | and it works perfect |
| 04:39.42 | Silik0n | juggie: I have seen x-lite be stupid |
| 04:39.55 | Juggie | id like to know why it was ignoring the sdp header |
| 04:39.55 | Deryl | it was notified to the freebsd list, I only run freebsd, so i only care about freebsd |
| 04:39.56 | *** part/#asterisk nolan-- (~chatzilla@216.199.159.79) |
| 04:40.03 | Juggie | and picking its own port |
| 04:40.04 | nwhit | its weird... i get that horrible call quality whenever this one phone touches the call |
| 04:40.17 | Mavvie | Deryl: you implied it by address the FreeBSD users and showing the FreeBSD security alert. |
| 04:40.17 | Juggie | broken phone |
| 04:40.23 | Nethab | i thought x-lite used 8000 |
| 04:40.30 | nwhit | even if that phone parks the call and another phone picks it up... the call never recovers |
| 04:40.33 | Silik0n | i have verified a possible bug where it always assumes NAT when its on RFC1918 address space and talking to a public IP space SIP Server |
| 04:40.33 | Nethab | which is bad enough |
| 04:40.45 | Deryl | Mavvie: YOU thought that. people running freebsd know how to take it. |
| 04:40.56 | nwhit | Juggie, so you think that it is a bad phone |
| 04:41.08 | Mavvie | Deryl: try finger edwin@freebsd.org and try again. |
| 04:41.14 | nwhit | Juggie, people on some of the other phones have complained also |
| 04:41.25 | Juggie | Nethab, it does use 8000 for receiving rtp from * |
| 04:41.36 | Deryl | Mavvie: no need. you have a problem with what i posted.. tough shit |
| 04:41.40 | Juggie | but for sending to * it should use what the sdp says |
| 04:41.48 | Juggie | but it doesnt unless you change that setting |
| 04:42.05 | Mavvie | Deryl: you're a big boy. Now have a lollipop and go happy playing in the corner. |
| 04:42.11 | Sedorox | ... |
| 04:42.15 | *** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net) |
| 04:42.27 | Sedorox | Delvar: Mavvie your both children.. so sit down and shut up :p |
| 04:42.44 | Pete_Largo | busy channel! |
| 04:42.57 | Mavvie | :-) |
| 04:43.04 | Sedorox | :p |
| 04:43.04 | Pete_Largo | Hi, I just found asterisk.org and thought it looked interesting |
| 04:43.09 | Deryl | Mavvie: if you are the same guy you purport to be, next time have clue enough to say hey just so you know.. this affects more than just XYZ. especially since the kerneltrap and other listings don't make that clear |
| 04:43.56 | Deryl | his fuckin problem if he has a problem with it |
| 04:45.19 | ManxPower | Silik0n: Not a bug. use nat=never to NEVER assume nat. |
| 04:45.44 | Silik0n | ManxPower: its not in asterisk its a problem in X-Lite |
| 04:45.57 | ManxPower | Silik0n: softphones suck. |
| 04:46.11 | mmlj4 | hey ManxPower: i got * up and running here, 4 softphones, they all work locally, and voicemail works # i think you were away the first time I posted this |
| 04:46.12 | Silik0n | yeah well i'm not lugging a hardphone everywhere i go |
| 04:46.15 | Nethab | i think softphones are soft and cuddly |
| 04:46.25 | Nethab | children are great, they taste just like chicken |
| 04:46.56 | nwhit | does anyone know where to find the exten little tags (template) for the snom phones? i've see it once and can't find it again |
| 04:47.18 | Silik0n | y0 kram |
| 04:47.39 | newmedian | Pete_Largo: welcome to our chaos. |
| 04:47.40 | ManxPower | mmlj4: See #$customer_city |
| 04:47.49 | mmlj4 | yep |
| 04:48.00 | Pete_Largo | Thanks newmedian |
| 04:48.38 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:48.48 | Juggie | ManxPower, nat=never had no effect |
| 04:48.57 | Juggie | its xlite, not asterisk |
| 04:49.03 | Juggie | sjphone worked fine |
| 04:49.19 | Silik0n | Juggie: what is the layout? |
| 04:49.35 | Silik0n | network wise wonder if you are seeing the same thing I am |
| 04:49.43 | Juggie | softphone<-nat->internet<-nat/dmz->asterisk |
| 04:50.13 | Silik0n | thats like begging for 1 way audio |
| 04:50.29 | ManxPower | Juggie: softphones suck. |
| 04:50.37 | Juggie | ManxPower, i'm not disputing that |
| 04:50.51 | ManxPower | Silik0n: you don't have the same RFC address space on both sides, do you? |
| 04:51.09 | Silik0n | ManxPower: nope |
| 04:52.05 | Juggie | Silik0n, sip is fine, if you understand the protocol |
| 04:52.14 | Silik0n | ManxPower: I'm seeing RFC1918 - router (NOT NAT) - asterisk (on non 1918 space) X-Lite ALWAYS assumes NAT and crazyness insues |
| 04:52.14 | Juggie | and you have control of the important parts of the network |
| 04:52.45 | Silik0n | Juggie: most people dont set it up right and end up with 1way audio on double nat like that tho |
| 04:52.48 | Juggie | xlite did the same thing for me on a private ip and public |
| 04:53.07 | Juggie | used 8000 and some 3000ish port |
| 04:53.37 | Silik0n | i did some debugging on the packets and X-lite was putting a pub IP in the sip From: |
| 04:53.39 | Juggie | it does the same thing when you force firewall to open ip |
| 04:53.57 | Juggie | thats easily solved, just set nat=yes |
| 04:54.04 | Juggie | doesnt matter what xlite puts in there |
| 04:54.23 | Silik0n | yeah but X-Lite still is broken |
| 04:54.33 | Silik0n | thats just a work around not a fix |
| 04:54.42 | Juggie | my problem was, in the sdp asterisk would say use 11100 or something as the rtp port |
| 04:54.53 | Juggie | and xlite would go ahead and use 3581 or something |
| 04:54.57 | Silik0n | heh |
| 04:55.04 | Juggie | now of course, my firewall is setup to forward 10000-13000 into asterisk |
| 04:55.08 | Juggie | because those are the rtp ports |
| 04:55.18 | Juggie | i have set in rtp.conf |
| 04:56.16 | Juggie | theres an option that says "obey reverse udp mapping rules" |
| 04:56.22 | Juggie | in advanced conf, in rtp and sip |
| 04:56.32 | Juggie | turn to no, and it now acts properly |
| 05:04.47 | firestrm | anyone know with nufone, if im only doing outbound calls do i need to do register => username:password@gw-sfld.nufone.net |
| 05:05.02 | firestrm | or is that for inbound only? |
| 05:05.31 | ManxPower | registration is only EVER for inbound |
| 05:06.02 | firestrm | ManxPower, thanks, thats what i thought, but i wasnt quite sure.. |
| 05:11.40 | *** part/#asterisk james_ed (~james_ed@ip68-103-200-171.ks.ok.cox.net) |
| 05:15.59 | JerJer | firestrm: use switch-2.nufone.net - you should have received and email quite a while ago asking you to update your config |
| 05:16.38 | Qwell | firestrm: finally got everything going? |
| 05:18.51 | Qwell | JerJer: y0 |
| 05:25.38 | PTG1234 | anyone in here set up ser before? :) |
| 05:25.40 | PTG1234 | hey qwell |
| 05:25.59 | Qwell | PTG1234: afternoon |
| 05:28.33 | Deryl | I wish NuFone would get another allocation of DIDs. I'd like to get another DID from them so I have one for each business venture. |
| 05:28.38 | firestrm | Call rejected by 66.225.202.72: No authority found.. that meas something wrong with username/password? |
| 05:33.28 | JerJer | where does it say 'NuFone support channel' ? |
| 05:33.40 | Qwell | JerJer: tattoo'd on your forehead I'm afraid |
| 05:34.50 | Silik0n | <PROTECTED> |
| 05:35.53 | firestrm | JerJer, why the crappy attitude? what did i ever do to you? Ive allways tryed to treat you with respect and gratitude, and in response you return the favor with childish snide remarks. What gives? |
| 05:36.23 | firestrm | im sorry is im not leet enough for you.. |
| 05:37.07 | *** join/#asterisk rcam (~rcammobil@adsl-218-151-77.jax.bellsouth.net) |
| 05:40.31 | Juggie | how do you change the sip settings in pulver communicator? do you have to buy it? |
| 05:41.43 | firestrm | Juggie, its in there, cant remember exactly but it was buried deep.. |
| 05:41.57 | firestrm | sorry i cant be more help.. |
| 05:42.17 | Juggie | it says to consult the site |
| 05:42.21 | Juggie | its like a game :) |
| 05:42.23 | Juggie | i'm digging |
| 05:42.54 | firestrm | ya i know.. it took me 2 hours to find it as well.. unfortunatly ive uninstalled it since.. |
| 05:45.13 | *** join/#asterisk McUnixJr (~mere@McUnixJr.gold.supporter.pdpc) |
| 05:46.01 | *** join/#asterisk slamb (~slamb@adsl-69-230-8-158.dsl.pltn13.pacbell.net) |
| 05:46.22 | Juggie | grrr |
| 05:46.38 | McUnixJr | ruff |
| 05:49.15 | firestrm | Juggie, i know how you feel.. im having issues with iax myself right now.. |
| 05:50.28 | Juggie | i just want to find the stupid key thats hidden :) |
| 05:50.38 | Qwell | no config file? |
| 05:50.43 | McUnixJr | what key? |
| 05:50.47 | McUnixJr | hey Qwell |
| 05:51.31 | slamb | Hey, does anyone here use a softphone on OS X? |
| 05:51.49 | McUnixJr | slamb, i use SJPhone |
| 05:52.00 | Juggie | McUnixJr, the key for pulver communicator is hidden on their site |
| 05:52.02 | Juggie | you have to find it |
| 05:52.16 | McUnixJr | ah - havent tried that |
| 05:52.26 | slamb | McUnixJr: Cool. How's audio quality / what codec do you use? I'm getting horrible distorting with X-lite. |
| 05:52.55 | McUnixJr | the default gsm and ulaw that comes with it |
| 05:52.56 | TheEmperor | does anyone know when i connect a pri modem to a balun and then from there to the * server e1 card, do i use a crossover cable or a straight through cable? |
| 05:53.39 | McUnixJr | audio quality is ok - some tinnish sounds |
| 05:54.13 | McUnixJr | i also use SJPone on my dell inspiron 700m, its really bad from that system for some reason. i think the mic is to close to the speakers and wind up getting feedback |
| 05:54.44 | nwhit | hmmm snom firmware 3.60i breaks intercom... not nice |
| 05:54.49 | *** join/#asterisk |neuro| (~|neuro|@212.176.51.231) |
| 05:54.54 | Qwell | Juggie: key to do what? |
| 05:55.06 | Juggie | change it from fwd to anothger sip server |
| 05:56.33 | slamb | McUnixJr: I'm giving SJPhone a spin on my Mac now. |
| 05:57.25 | *** part/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 05:57.33 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 06:02.49 | Juggie | bah |
| 06:03.48 | Qwell | are they even selling it yet? |
| 06:04.20 | slamb | McUnixJr: Huh. I can get it to connect and make a call. I hear the ringing noise. I see from my Asterisk logs that it's responding to DTMF input. But I don't get any audio at all through SJphone. |
| 06:06.32 | firestrm | if im getting a No authority found, does this mean that the server is rejecting my Username/pass? or could there be soemthing else affecting this? |
| 06:07.09 | Qwell | Juggie: It isn't free... |
| 06:07.19 | Juggie | i cant find where to buy it |
| 06:07.34 | Juggie | and all the manual says is "you have to procure a key" |
| 06:08.07 | firestrm | Juggie, if you need to buy it, i wouldnt go with pulver communicator.. there are much better softphones out there. |
| 06:08.31 | slamb | firestrm: IIRC, that means it's looking for a user name that doesn't exist. (Or a peer name? I don't remember which is incoming and which is outgoing.) |
| 06:08.47 | slamb | firestrm: for example, I got that error when my iax.conf had a [sixtel] section instead of [sixTel] |
| 06:08.57 | Juggie | i thought you said you found it on their site |
| 06:10.19 | firestrm | slamb, hmmm ive checked and double checked the iax.conf.. its cut n paste from the "nufone" setup page on voip-info.. |
| 06:11.25 | Juggie | bah |
| 06:11.58 | slamb | firestrm: hmm. maybe "iax2 debug" will show you what it's looking for. |
| 06:13.40 | firestrm | slam: debug is not much help.. username is correct, server is correct.. i know for certan im using the correct pass.. its just not allowing the connection, like the nufone machine doesnt recognise me.. |
| 06:13.59 | TheEmperor | so anyone know if I need to use a crossover cable or a straight through cable to connect the pri to the e1 card? |
| 06:15.00 | firestrm | TheEmperor, straight, only use crossover from card to card on different machine without hub/switch/router between |
| 06:15.02 | slamb | firestrm: hmm. afraid I can't help more, then. I've been using Asterisk for 24 hours now, so... |
| 06:15.18 | Qwell | firestrm: Thats for networking :p |
| 06:15.20 | TheEmperor | firestrm: ah! thank you :) |
| 06:16.05 | firestrm | slamb: thanks for trying, at least your attitude is better than "those who are too leet" for us unwashed normals |
| 06:16.27 | firestrm | TheEmperor: no prob.. |
| 06:17.04 | Qwell | firestrm: When does it error? |
| 06:17.11 | Qwell | on register? |
| 06:19.13 | firestrm | Qwell, im not going inbound on it , so im not registering.. when i try outbound i get Call rejected by 66.225.202.72: No authority found |
| 06:19.34 | *** join/#asterisk BoRiS (~boris@wnpgmb01dc2-25-225.dynamic.mts.net) |
| 06:19.37 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 06:19.39 | Qwell | pastebin the relative sections of your config(s) |
| 06:19.44 | Qwell | passwordless, of course |
| 06:19.47 | firestrm | :) |
| 06:20.02 | firestrm | ok.. working on it.. |
| 06:20.28 | Qwell | post the link here when you're done. brb |
| 06:20.46 | *** join/#asterisk roamer323 (~sing@Toronto-HSE-ppp3681114.sympatico.ca) |
| 06:23.46 | PTG1234 | anyone here use ser with asterisk? :) |
| 06:24.49 | firestrm | Qwell: http://pastebin.ca/11650 |
| 06:25.44 | blitzrage | oh the joys of PHP and PostgreSQL! |
| 06:27.30 | *** join/#asterisk remmo (~rem@smack.isp.net.au) |
| 06:27.43 | remmo | i hate networks and css |
| 06:31.19 | Qwell | firestrm: looks fine to me...pretty much exactly what I have |
| 06:32.11 | firestrm | blitzrage: i feel for you.. i personally feel that the only way PHP makes sense if if your taking PCP :P |
| 06:32.43 | remmo | if ppl can not understand PHP then they should not be programming |
| 06:33.13 | Qwell | firestrm: you don't have a second type=peer for that or something, do you? |
| 06:33.20 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 06:34.03 | firestrm | Qwell, within the nufone definition or within the iax.conf anywhere? |
| 06:34.12 | Qwell | iax.conf |
| 06:34.38 | Qwell | I don't really know enough to be able to tell you what the problem might be |
| 06:35.17 | firestrm | Qwell: i do for my [iaxfwd] def, but could that really be causing the problem? |
| 06:35.42 | Qwell | no, I mean another [NuFone] peer or something silly |
| 06:35.59 | firestrm | no, nothing at all.. |
| 06:36.02 | firestrm | just the one.. |
| 06:37.43 | firestrm | Qwell: i suspect that the problem is not on my side.. (im not going to say more about that one for fear of invoking the wrath of jerjer) |
| 06:38.20 | BoRiS | Anyone have a newer beta firmware for the Senao SI-7800 wifi phone then 10/17/2004? msg me please |
| 06:39.10 | BoRiS | hehe |
| 06:39.21 | *** part/#asterisk |neuro| (~|neuro|@212.176.51.231) |
| 06:40.39 | firestrm | BoRiS: ever seen these? http://www.abptech.com/mainpages/products/HCL-WirelessIP5000.html |
| 06:41.03 | firestrm | BoRiS: they look quite sexy as far as wifi sip phones go.. |
| 06:41.12 | BoRiS | Yeah....someone here in the channel has one of them.... :) |
| 06:41.40 | firestrm | BoRiS: any reports from them? im thinking about buying a bunch for a project.. |
| 06:42.58 | ManxPower | Mozilla Calendar would be a lot nicer if it supported printing. |
| 06:45.22 | JerJer | ok who wants to be the first to send a payment in and have it automatically apply to their nufone account? |
| 06:46.26 | firestrm | JerJer: i would be happy if your server accepted my username |
| 06:46.52 | Wonka | .oO( ouch ) |
| 06:46.57 | firestrm | JerJer: but that does sound line a nice accomplishment |
| 06:49.51 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 06:50.51 | JerJer | if you expect me to help you i will at the very least need your username |
| 06:51.01 | firestrm | Wonka: i know.. i could have been a bit more tactful.. sorry jarjar |
| 06:51.35 | Wonka | ;) |
| 06:51.45 | Zeeek | Hey JerJer |
| 06:52.54 | Zeeek | JerJer is there a way to know what preferred codecs are stored for national tollfree nufione numbers? |
| 06:53.21 | Zeeek | I see when you order a new one, you can specify this (when there are new ones offered, that is) |
| 06:54.06 | JerJer | Zeeek: nope - that option is highly broken |
| 06:54.28 | JerJer | it will be fixed with the new version that will launch here someday |
| 06:54.51 | JerJer | only if you run an IAXy should you really be setting a codec restrction |
| 06:55.12 | JerJer | cuz u can allow whatever codec's you want with asterisk |
| 06:55.15 | Zeeek | Ok, I never figured out why I had that g729 problem with those DID |
| 06:55.37 | JerJer | some do pass-thru with their DIDs as well and want a codec forced |
| 06:55.45 | Zeeek | It was because of the license problem, but only the DID cause the symptom by seeming to insisit on g729 |
| 06:55.46 | Qwell | JerJer: paypal or cc? |
| 06:56.04 | JerJer | same same |
| 06:56.19 | JerJer | you can give a CC to paypal if you want - without having a paypal account |
| 06:57.01 | Zeeek | paypal has one advantage over paying directly by cc: they don't care about the shippiong address |
| 06:58.34 | *** join/#asterisk n4y (~frodo7@host-ip226-209.crowley.pl) |
| 06:59.57 | firestrm | JerJer: congrats on the new functionality.. i look forward to using it. its just was was needed :) |
| 07:00.16 | Sato1 | yes!! |
| 07:00.19 | Sato1 | oops |
| 07:00.20 | JerJer | someone without a paypal account and a credit card try it - the only cc i have is tied to the paypal account and they are being bitches about letting me use it |
| 07:01.21 | firestrm | JerJer: i didnt know paypal would let you do that.. even better :) |
| 07:01.52 | ManxPower | What is the link to send money via paypal without having an annount on paypal? |
| 07:01.59 | JerJer | we had to do a bit of extra integration to make it happen |
| 07:02.17 | JerJer | it = credit cards without paypal accounts |
| 07:02.20 | JerJer | -s |
| 07:05.59 | *** join/#asterisk darby_t (~tom@host-ip226-209.crowley.pl) |
| 07:06.15 | syle | lol credit cards are the most important |
| 07:06.26 | syle | how many people who don;t use internet know what paypal is |
| 07:06.32 | syle | but everyone knows what a credit card is |
| 07:06.52 | Qwell | how many potential voip customers don't know what paypal is? |
| 07:07.15 | syle | if i was to guess 50% |
| 07:07.23 | Qwell | I'd say more like 95% |
| 07:07.39 | syle | i think you give more credit than is due |
| 07:07.44 | Qwell | perhaps |
| 07:07.59 | syle | your not a programmer i guess |
| 07:08.10 | Qwell | users != customers |
| 07:08.22 | Qwell | and, I am |
| 07:08.37 | Qwell | users are idiots |
| 07:08.45 | Qwell | customers are still idiots, but they pay you |
| 07:08.47 | syle | your not showing that then are you :) |
| 07:08.51 | Nethab | everyone is a potential customer |
| 07:09.06 | Nethab | not all customers are l33t |
| 07:10.03 | syle | here is my question about voip, say you open a company using asterisk+ser or whatever, you charge 20 bucks a month unlimited, what are your real expenses? bandwidth, server, isn;t there a charge to get them a phone number and billed to you per month as well? |
| 07:10.55 | firestrm | Qwell, yes true, but if they arent smart enough to know what paypal is, will they really be setting * boxen, and given that what are the chances they will be using nufone without asterisk? |
| 07:11.13 | JerJer | yeah you have to be pretty well informed to find nufone |
| 07:11.17 | JerJer | that's for damn sure |
| 07:11.30 | JerJer | try finding us in search engines without using our name |
| 07:12.00 | Qwell | firestrm: exactly my point |
| 07:12.01 | syle | you run nufone website jerjer? |
| 07:12.05 | firestrm | JerJer: tell me about it, im an intermediate * user, and i still have the bruses to prove it :) |
| 07:12.29 | JerJer | syle: i own and run nufone |
| 07:12.42 | syle | good i been looking to talk to you , PM |
| 07:14.03 | blitzrage | I can't believe i'm still up coding |
| 07:14.08 | blitzrage | someone shoot me :) |
| 07:14.17 | Qwell | blitzrage: I'll tell you a little story instead. |
| 07:14.20 | blitzrage | *gasp* |
| 07:14.26 | blitzrage | Qwell: please |
| 07:14.32 | blitzrage | syle: oh you have no idea :) |
| 07:14.34 | ManxPower | syle has style. |
| 07:14.37 | blitzrage | syle: I hate running out |
| 07:14.37 | Qwell | once upon a time, a young programmer was up all night coding to his hearts content |
| 07:14.47 | Qwell | the next morning, shit was b0rked |
| 07:14.53 | syle | yeah but its a pain to grow your own |
| 07:14.54 | Qwell | young programmer no longer codes after midnight |
| 07:15.03 | blitzrage | Qwell: I'm more careful when I'm this tired because I know how tired I am :) |
| 07:15.12 | Qwell | blitzrage: Thats what I told myself too. |
| 07:15.14 | blitzrage | its 3:15am here :) |
| 07:15.18 | blitzrage | Qwell: lol |
| 07:15.23 | blitzrage | Qwell: so far, everything is working :) |
| 07:15.23 | Qwell | erm... |
| 07:15.27 | Qwell | thats what he told himself |
| 07:15.39 | blitzrage | Qwell: haha, stay in character :) |
| 07:15.45 | Qwell | yeah, its not about me |
| 07:15.59 | firestrm | blitzrage, try assembler with no sleep.. i used to code embedded apps for avionics till 4am.. |
| 07:16.03 | syle | all night coding sessions consist of downing 2L of pepsi or coke, a pack of cigarrettes for frusteration and a few joints for tolerance |
| 07:16.15 | syle | i know the game quite well :) |
| 07:16.51 | Qwell | ManxPower: I thought you didn't code? |
| 07:17.00 | Qwell | or, is that a subtle hint that you're never sober? :) |
| 07:17.06 | ManxPower | Qwell: "code" is such a broad term. |
| 07:17.12 | JerJer | mmmm joints |
| 07:17.26 | firestrm | lol |
| 07:17.27 | blitzrage | aye |
| 07:17.45 | ManxPower | Qwell: I call it code, others call it "abomination". |
| 07:17.51 | Qwell | ahh... |
| 07:17.53 | blitzrage | I rarely program - but now I'm the main Asterisk admin for a service provider, so thus, I code |
| 07:18.23 | blitzrage | I think better when I smoke joints :) |
| 07:18.26 | ManxPower | Qwell: I write SMALL utilities, like the outcall notify and group voicemail and stuff like that. |
| 07:18.34 | Qwell | ManxPower: I see |
| 07:19.03 | blitzrage | never smoked in highschool r first year of college - first year of college, got C's and D's - started smoking... grades went up to B's and A's - go figure |
| 07:19.14 | blitzrage | ManxPower: wish I had beer too |
| 07:19.28 | blitzrage | really gotta send out some invoices I guess ;) |
| 07:19.29 | JerJer | mmmmmm beeeer |
| 07:19.38 | blitzrage | JerJer: what are you still doing up? :) |
| 07:19.47 | JerJer | coding like a little bitch |
| 07:19.48 | ManxPower | blitzrage: the last time I mixed alcohol and pot I passed out on someone's bathroom floor. I don't do that anymore. |
| 07:19.53 | Qwell | JerJer: You and Greg hitting cluecon? |
| 07:20.11 | blitzrage | ManxPower: you just need to know how to balance them - don't ever smoke after you're drunk - bad news all around |
| 07:20.21 | ManxPower | blitzrage: *nod* |
| 07:20.22 | blitzrage | weed before beer - in the clear :) |
| 07:20.38 | blitzrage | ManxPower: smoke after you're drunk, and you might as well just put yourself in the gravitron |
| 07:20.38 | ManxPower | I don't drink much anymore anyway. Bad liver. |
| 07:20.53 | blitzrage | ahhhh, crappy. I don't drink much to begin with |
| 07:21.11 | ManxPower | blitzrage: I prefer...other...intoxicants anyway. |
| 07:21.25 | JerJer | Qwell: not if they expect us to pay 700 bucks |
| 07:21.29 | JerJer | plus hotel |
| 07:21.30 | JerJer | plus beer |
| 07:21.41 | JerJer | plus dinners |
| 07:21.44 | blitzrage | ManxPower: heroin? |
| 07:21.46 | blitzrage | :) |
| 07:21.58 | JerJer | i dont mind Beer, Hotel, Dinner |
| 07:22.01 | ManxPower | blitzrage: LOL! That's one of the few things I've not at least tried. |
| 07:22.34 | *** join/#asterisk valence (~valence@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 07:22.35 | Zeeek | have you tried the lord? |
| 07:22.36 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 07:22.38 | firestrm | blitzrage, lol |
| 07:22.41 | blitzrage | ManxPower: me either, tried cocain a couple of times just to see what all the hype was about, wasn't impressed. Thats about it other than weed and beer (how Canadian of me :)) |
| 07:22.47 | blitzrage | Zeeek: lol |
| 07:23.02 | ManxPower | Zeeek: Naw! That shit will fuck you up bad. |
| 07:23.06 | Zeeek | blitzrage a musician friend heard that from his mother in law |
| 07:23.16 | Zeeek | "you've tried everything else!" |
| 07:23.19 | blitzrage | Zeeek: lol |
| 07:23.37 | JerJer | good cocain can be fun |
| 07:23.46 | JerJer | but the bullshit that most have isn't worth it |
| 07:23.53 | firestrm | naw, i didnt like it.. felt like too much coffee |
| 07:24.11 | blitzrage | damn semi-colons will do it to you every time! |
| 07:24.13 | JerJer | I miss George Young |
| 07:24.13 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 07:24.16 | Zeeek | a sniff of coke makes you feel like a new man... trouble is, then the new man wants some |
| 07:24.24 | firestrm | lol |
| 07:24.35 | Zeeek | George Carlin |
| 07:24.46 | blitzrage | maybe I was too intoxiated, or it sucked ass all the times I did it, but honestly all I got from it was a numb throat |
| 07:24.57 | Zeeek | well, I've had the ultimate geek's nightmare |
| 07:25.04 | Zeeek | now I have to send my dream phone back, sniff |
| 07:25.07 | JerJer | blitzrage: it was prolly cut at least a few times then |
| 07:25.23 | ManxPower | Isn't this better talked about on #astrisk-drinkers? |
| 07:25.41 | Zeeek | but first, to the market, food for the body |
| 07:25.57 | Zeeek | oh ManxPower if you do come to Paris, be sure and let me know |
| 07:25.58 | JerJer | lamers cut the shit with viteman A and C, baby asprin or other worse shit |
| 07:26.13 | remmo | special K |
| 07:26.16 | Zeeek | lecithin |
| 07:26.19 | ManxPower | Zeeek: Well I have three days unplanned. |
| 07:26.20 | blitzrage | haha, here's the kind of code I'm creating: $status = "If I wasn't so tired at 3:21 AM, this would have worked"; |
| 07:26.21 | firestrm | Zeeek, what hapenned to yr fone |
| 07:26.25 | firestrm | ? |
| 07:26.30 | Zeeek | it died young |
| 07:26.37 | ManxPower | I extended my stays in Amsterdam and Amtwerp. |
| 07:26.47 | Zeeek | Amsterdam rocks |
| 07:26.53 | Zeeek | for a few days, anyway |
| 07:27.01 | Zeeek | already did all the rituals |
| 07:27.04 | blitzrage | no need to go to Amsterdamn, same shit in Canada ;) |
| 07:27.14 | ManxPower | Zeeek: Do you have any idea how long it takes to get from Amsterdam to Paris via train? Also Amsterdam to Madrid? |
| 07:27.14 | JerJer | BC! |
| 07:27.20 | blitzrage | JerJer: aye cap'n |
| 07:27.28 | firestrm | blitzrage, JerJer, BC = Best Cannabas |
| 07:27.35 | Zeeek | my guess would be about 8hs? |
| 07:27.36 | blitzrage | too bad I'm in Ontario :) |
| 07:27.38 | ManxPower | blitzrage: Canada does not have http://www.blacktulip.nl/ |
| 07:27.40 | Zeeek | 8 hours |
| 07:28.03 | blitzrage | ManxPower: lol - fine with me |
| 07:28.30 | ManxPower | blitzrage: Oddly, Toronto has something quite similar. |
| 07:28.35 | *** join/#asterisk TheEmperor (~TheEmpero@203.114.48.47) |
| 07:28.42 | ManxPower | We stayed there twice. |
| 07:28.55 | Zeeek | ManxPower Amst-Paris = about 5hrs |
| 07:28.57 | blitzrage | ManxPower: yah... probably on Church street right? |
| 07:29.10 | ManxPower | blitzrage: Just off Church. Called The Bent Inn |
| 07:29.18 | blitzrage | hahaha... how'd I guess, lol |
| 07:29.20 | Zeeek | ManxPower and anyone lese needing info : http://www.voyages-sncf.com/ |
| 07:29.22 | hypa7ia | and there's always the Hot Box Cafe |
| 07:29.32 | blitzrage | I could go for some hot box :) |
| 07:29.44 | blitzrage | saw some in the pool earlier in the night |
| 07:30.15 | blitzrage | well, that's it for me - off to bed! |
| 07:30.18 | blitzrage | night all |
| 07:30.41 | hypa7ia | ManxPower: torontonian? |
| 07:30.53 | ManxPower | hypa7ia: no, like to visit. |
| 07:31.03 | firestrm | me too.. got my nufone connection working.. im content and happy.. to sleep for me.. |
| 07:31.08 | hypa7ia | cool |
| 07:31.25 | ManxPower | If canada wasn't so cold I'd move there. |
| 07:31.30 | firestrm | thanks again JerJer! |
| 07:31.41 | hypa7ia | ManxPower: lol |
| 07:33.02 | TheEmperor | can someone please tell me what this means? WARNING[1185929152]: chan_zap.c:5993 zt_pri_error: PRI: !! Got S-frame while link down |
| 07:33.29 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 07:33.47 | implicit | hello |
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| 07:39.58 | nextime | anyone using pyastre? |
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| 07:43.09 | Newbie___ | MikeJ[Laptop]: thanks for yesterday |
| 07:43.23 | *** join/#asterisk oej (~oej@213.204.186.40) |
| 07:43.38 | BoRiS | Hey JerJer, How did you integrate paypal into your system for credit cards without having people to sign up? |
| 07:44.00 | BoRiS | :) |
| 07:44.31 | JerJer | I logged into paypal and RTFM |
| 07:44.37 | JerJer | for about 4 hours today |
| 07:45.05 | *** join/#asterisk Brook_Jiang (~brook@210.82.30.5) |
| 07:45.33 | Romik | jerjer: nice thing to read! |
| 07:49.42 | Newbie___ | anyone did a callback on * ? |
| 07:49.52 | JerJer | callback is trivial |
| 07:50.03 | JerJer | provided you have a reliable trigger |
| 07:51.10 | Newbie___ | JerJer: is it worth while to buy one off the shelf ? |
| 07:52.51 | JerJer | hell no - write one with asterisk |
| 07:54.13 | Newbie___ | JerJer: any good recommendation ? |
| 07:54.19 | JerJer | vi |
| 07:54.24 | JerJer | type for a few hours |
| 07:54.29 | JerJer | esc shift z z |
| 07:54.32 | JerJer | run it |
| 07:54.35 | Newbie___ | lol |
| 07:54.52 | Newbie___ | JerJer: i am not familiar with * |
| 07:55.11 | Newbie___ | enough headache with the present * box |
| 07:55.13 | JerJer | then pay somene (not me) to develop you an app |
| 07:55.36 | Newbie___ | know any one ? |
| 07:58.47 | newmedian | You may want to ask that question again during the daytime, or after dinner, EST. |
| 07:59.26 | Newbie___ | newmedian: thanks |
| 07:59.41 | newmedian | np. currently 4am EST. Of course, who needs sleep anyway. |
| 08:00.59 | Brook_Jiang | help |
| 08:03.54 | JerJer | what is this word sleep ? |
| 08:05.40 | shido6 | ZzzzzzZZZzz |
| 08:07.07 | [hC] | JerJer: i looked it up in my dictionary and it said "See ______" |
| 08:07.19 | [hC] | It looks like nobody has ever figured it out. |
| 08:13.06 | *** join/#asterisk bjohnson (~bjohnson@66.11.188.6) |
| 08:14.29 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 08:14.57 | Aze`DEV | Anyone have problem with cvs head and bristuff ? |
| 08:17.41 | *** part/#asterisk oej (~oej@213.204.186.40) |
| 08:19.36 | *** join/#asterisk oej (~oej@213.204.186.40) |
| 08:19.49 | rabelais | is doing a phone number transfer a custom order (something that can't be done from the website) from iax.cc? |
| 08:20.03 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 08:24.39 | *** join/#asterisk Savage-S (~savage@c514701e0.cable.wanadoo.nl) |
| 08:29.02 | *** join/#asterisk mbranca (~matteo@81.208.92.210) |
| 08:32.09 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 08:40.03 | *** join/#asterisk newl (~newlook@203-59-164-248.dyn.iinet.net.au) |
| 08:47.00 | *** join/#asterisk eper-werk (~eperdeme@telkom.gotadsl.co.uk) |
| 08:50.54 | *** join/#asterisk ChrisHodgetts (~chris@topanga.archnetnz.com) |
| 08:53.32 | ChrisHodgetts | does anyone know what this could mean? |
| 08:53.49 | ChrisHodgetts | May 13 20:13:56 WARNING[7864]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 2202927650@192.168.100.22 for seqno 21 (Non-critical Response) |
| 08:54.13 | oej | A SIP packet did not reach its target |
| 08:54.25 | oej | sip show channels will show you what's going on |
| 08:55.05 | ChrisHodgetts | the call setup was, SIP Phone > Asterisk > SIP Proxy > Outbound call |
| 08:55.30 | ChrisHodgetts | would direction would that packet have been droped in |
| 08:55.37 | oej | This could be something else as well, a qualification packet |
| 08:55.45 | oej | You need to turn on "sip debug" to see what's going on |
| 08:56.03 | ChrisHodgetts | arr ok |
| 08:56.11 | ChrisHodgetts | thanks for your help, I will go and do that and have a look |
| 09:00.07 | *** join/#asterisk n4y (~frodo7@host-ip226-209.crowley.pl) |
| 09:00.54 | eper-werk | has anybody used, voiptalk shop website to order things? |
| 09:03.45 | *** join/#asterisk vpp (~noone@host-83-146-50-131.bulldogdsl.com) |
| 09:03.51 | vpp | hi |
| 09:11.40 | *** join/#asterisk jeffgus (~jeffgus@alpha.zimage.com) |
| 09:19.49 | vpp | anyone around |
| 09:20.35 | *** join/#asterisk jeffgus (~jeffgus@alpha.zimage.com) |
| 09:22.44 | *** join/#asterisk Martohtar (Martohtar@82.196.218.80) |
| 09:22.48 | *** join/#asterisk tiko_007 (~tiko_007@218.108.174.21) |
| 09:22.59 | *** join/#asterisk _omer (~dfsdf@202.147.174.178) |
| 09:24.02 | _omer | anybody there? |
| 09:24.44 | vpp | hi |
| 09:25.32 | _omer | vpp: what should I type in linux shell to go back to GUI |
| 09:25.33 | _omer | ? |
| 09:25.35 | _omer | :( |
| 09:26.59 | vpp | how did u get to the linux shell? |
| 09:27.35 | _omer | when I started my machine....I got the shell |
| 09:27.44 | vpp | oh |
| 09:27.48 | vpp | well then u can try |
| 09:27.50 | vpp | x11start |
| 09:27.58 | _omer | its red hat linux |
| 09:28.10 | vpp | or startx |
| 09:28.30 | vpp | did u install X on it? |
| 09:28.37 | _omer | startx ....works....thanks |
| 09:28.50 | _omer | yes.....now I'm in x :) |
| 09:28.50 | vpp | ok np |
| 09:30.05 | *** join/#asterisk th (~th@montana.hbsn.de) |
| 09:31.09 | th | i'm searching for commercial support in germany. anyone here to offer such? |
| 09:31.41 | oej | Isn't germany a southern part of Sweden? If so, we're available... |
| 09:31.58 | th | heh |
| 09:32.58 | vpp | LOL |
| 09:34.36 | t0p | hi,what is the 20 and tr in Dial(Zap/1/${EXTEN},20,tr) for? I waited for 20 secs but it didn't jump to the next step |
| 09:36.20 | *** join/#asterisk rsdvd (~rsdvd@rsdvd.plus.com) |
| 09:37.42 | *** join/#asterisk Jas_Williams (~Jason@host217-43-100-176.range217-43.btcentralplus.com) |
| 09:38.33 | t0p | Jas_Williams: Hi |
| 09:39.12 | Jas_Williams | Morning fixed your X100P yet |
| 09:39.34 | t0p | Jas_Williams: yeah, that card is broken |
| 09:39.52 | vpp | anyone know why asterisk says 'congested' when trying to dial out of Zap/g1 when it boots up |
| 09:39.54 | Jas_Williams | At least you now know :) |
| 09:40.01 | t0p | Jas_Williams: it looks like a new card but it didn't work |
| 09:40.04 | vpp | but if i do reload (twice) it works |
| 09:40.04 | vpp | ?! |
| 09:40.17 | t0p | Jas_Williams: thanks again for your help yesterday |
| 09:40.59 | Jas_Williams | No prob. |
| 09:42.24 | Jas_Williams | vpp: what zap card do you have in your server ? |
| 09:42.48 | t0p | Jas_Williams: what types of (other) cards are you using now? |
| 09:43.19 | vpp | Jas_Williams: Sangoma |
| 09:43.33 | vpp | T1 card |
| 09:43.42 | t0p | Jas_Williams: I am thinking of purchasing the card with 4 x E1s interface but it's pretty costly |
| 09:44.39 | Jas_Williams | t0p: I use an X100P in my lab and a Frtiz PCI BRI in small office PBX and 4 port E1 card in large office PBX |
| 09:44.40 | t0p | vpp: cool, for own use or for a business? |
| 09:45.26 | t0p | Jas_Williams: i see. is it an E1 CAS? |
| 09:45.29 | vpp | t0p: just testing it out at the moment |
| 09:45.30 | Jas_Williams | lab (HOME) Small PBX my Office Large office client install |
| 09:46.13 | Jas_Williams | t0p: No asterisk does not support E1 cas over ZAP I'm using E1 PRI Q931 EuroISDN |
| 09:46.19 | *** join/#asterisk HIValentine (jak@defiant.ircii.org) |
| 09:46.30 | HIValentine | hi |
| 09:46.35 | HIValentine | anybody use asterisk on bsd? |
| 09:47.14 | t0p | Jas_Williams: I thought I saw a configuration for E1 CAS somewhere. let me have a look |
| 09:47.33 | *** join/#asterisk atporter (atporter@disorder.primate.net) |
| 09:48.09 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 09:48.30 | Mc_Tr | hi! |
| 09:48.32 | Xavvy | hi |
| 09:48.48 | Mc_Tr | anoybody use pre-paid application? |
| 09:49.13 | Mc_Tr | i see this page: http://voip-info.org/wiki-CallingCard+Applications |
| 09:49.44 | vpp | anyone have an ideas? |
| 09:50.17 | Mc_Tr | sorry, vpp, no idea |
| 09:50.46 | zoa | we have a prepay and post pay application |
| 09:50.48 | zoa | but no website |
| 09:51.21 | Xavvy | so nobody here runs asterisk on bsd? |
| 09:51.47 | zoa | oh yes they do |
| 09:51.48 | zoa | i did once |
| 09:51.52 | zoa | but not any more |
| 09:51.57 | zoa | seems to work fine on bsd |
| 09:52.00 | Xavvy | hehe that doesn't sound good :) |
| 09:52.08 | zoa | just no one who used bsd and asterisk is awake now |
| 09:52.13 | Xavvy | well |
| 09:52.16 | Xavvy | you'll do fine |
| 09:52.31 | Xavvy | I just want to know your opinion on its stability |
| 09:52.34 | vpp | i read there were problems. so i used centos |
| 09:52.38 | vpp | but those were pretty old posts |
| 09:52.45 | Xavvy | yeah I heard things as well |
| 09:52.49 | Xavvy | which is why I came here :P |
| 09:52.52 | vpp | heheh |
| 09:53.02 | vpp | well i use centos 3.4 and its running fine |
| 09:53.23 | Xavvy | well, you said you used to run it on bsd |
| 09:53.23 | t0p | Jas_Williams: It's at http://pastebin.ca/11654 |
| 09:53.26 | Xavvy | it worked fine for you? |
| 09:53.39 | t0p | Jas_Williams: I don't know if it will work |
| 09:54.09 | vpp | i think there's alot more people running it on redhat |
| 09:54.16 | Xavvy | yeah |
| 09:54.21 | vpp | (centos is modified redhat enterprise) |
| 09:54.27 | Xavvy | I don't like redhat, though :) |
| 09:54.34 | vpp | i wanted to use slackware :p |
| 09:54.43 | _omer | I dont like linux :) |
| 09:54.43 | Xavvy | slack is awesome |
| 09:54.51 | vpp | yup :) |
| 09:54.53 | Xavvy | when I ran linux I started out with slack |
| 09:55.02 | Xavvy | a buddy of mine even owns darkstar.frop.org now, hehe |
| 09:55.07 | vpp | but in the end i thought there's no point being one of few running on that dist |
| 09:55.09 | Xavvy | but I switched to gentoo |
| 09:55.13 | Xavvy | (before switching to bsd) |
| 09:55.14 | vpp | the problem is when u ask for support.. even in here |
| 09:55.23 | Xavvy | well |
| 09:55.26 | vpp | most people blame the dist, or say oh i don't know i don't run it on that |
| 09:55.36 | Xavvy | any distro-specific problems I can ninja out myself |
| 09:55.52 | vpp | really your not gonna use the asterisk machine for anything else |
| 09:55.57 | Xavvy | I'm just wanting to know whether I should run it on bsd or not |
| 09:56.00 | vpp | it'll just sit there handling your calls hehe |
| 09:56.08 | Xavvy | yeah |
| 09:56.10 | vpp | u just wanna use sip? |
| 09:56.11 | Jas_Williams | t0p: this is using unicall a new library from www.soft-switch.org |
| 09:56.14 | vpp | iax etc? |
| 09:56.16 | Xavvy | yup |
| 09:56.19 | vpp | the built in stuff? |
| 09:56.23 | vpp | then i'm sure it will be fine |
| 09:56.26 | Xavvy | no hardware involved |
| 09:56.29 | *** part/#asterisk rsdvd (~rsdvd@rsdvd.plus.com) |
| 09:56.30 | *** join/#asterisk eivindtr (~eivindtr@062016241059.customer.alfanett.no) |
| 09:56.33 | Xavvy | unless I start doing faxes |
| 09:56.35 | vpp | if u want to use h323/oh323 then stick to centos or redhat |
| 09:56.42 | Xavvy | which is a whole different ballpark I hear |
| 09:56.56 | Xavvy | do you use faxes? |
| 09:57.00 | vpp | well as long as your using asterisk core stuff i don't think there will be a problem |
| 09:57.00 | Jas_Williams | t0p: It should work if you follow these notes http://www.soft-switch.org/unicall/installing-mfcr2.html |
| 09:57.12 | atporter | anyone got an iax<->pstn provider they're happy with? |
| 09:57.16 | vpp | i use it for sip -> h323 or sip -> PRI or h323 -> pri |
| 09:57.31 | vpp | basically to translate |
| 09:57.40 | vpp | its a cheap way to do it |
| 09:57.46 | Xavvy | ah |
| 09:57.53 | t0p | Jas_Williams: Okay, is the library free? |
| 09:58.09 | Jas_Williams | t0p: yes its open source |
| 09:58.18 | vpp | i ended up using centos because i found the asterisk @ home cd |
| 09:58.27 | vpp | now i boot that |
| 09:58.44 | vpp | then move atserisk, libpri, zaptel and download the cvs ones |
| 09:58.49 | vpp | then switch off all the crap |
| 09:58.51 | vpp | and i'm done |
| 09:58.56 | vpp | install in 20 mins :p |
| 09:58.59 | Sato1 | hiya vpp |
| 09:59.11 | vpp | hey Sato1 |
| 09:59.57 | vpp | Sato1: i'm trying asterisk with a T1 card now :) |
| 10:00.10 | vpp | brb |
| 10:00.58 | Sato1 | great! |
| 10:01.03 | Sato1 | good luck |
| 10:01.44 | Jas_Williams | t0p: The author of the unicall library is quite often here |
| 10:01.50 | Jas_Williams | ~seen coppice |
| 10:01.51 | jbot | coppice <~chatzilla@43.198.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 1d 18h 27m 18s ago, saying: 'ariel_: goodnight'. |
| 10:03.34 | t0p | Jas_Williams: i see |
| 10:04.56 | t0p | Jas_Williams: I have 10 E1 Interfaces here with obsolete dialogic voice boards |
| 10:05.50 | Jas_Williams | t0p: Some dialogic boards are supported by asterisk however you need to purchace a license from digium to use these boards |
| 10:06.26 | t0p | Jas_Williams: yeah but it's too bad that the ones I am using aren't supported |
| 10:06.31 | vpp | back |
| 10:06.38 | vpp | Sato1: thanks |
| 10:06.45 | vpp | having minor problems right now |
| 10:06.50 | vpp | but otherwise its working |
| 10:07.04 | *** join/#asterisk DannyF (~dannyf@h197n2fls32o865.telia.com) |
| 10:07.38 | t0p | Jas_Williams: thinking of changing from MFC/R2 to ISDN |
| 10:08.24 | t0p | Jas_Williams: what drivers/libraries are you using for ISDN? only Zaptel? |
| 10:08.30 | Jas_Williams | t0p: ISDN is better as all signalling is out of band rather than dtmf tones for dinis ANI etc |
| 10:10.05 | *** join/#asterisk Vercingetorix (~icechat5@69-173-140-135.agstme.adelphia.net) |
| 10:10.09 | t0p | Jas_Williams: is TE405P you are using? |
| 10:10.30 | Jas_Williams | t0p: Yes Thats the one |
| 10:10.45 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 10:12.09 | Vercingetorix | Hi, could someone tell me how to checkout CVS HEAD? I need the cvs checkout command line syntax. Thanks. |
| 10:12.44 | Jas_Williams | cvs checkout zaptel libpri asterisk |
| 10:12.50 | Vercingetorix | Putting dog out -- BRB. |
| 10:13.25 | Zeeek | HEAD won't work if you have a dog |
| 10:13.41 | Zeeek | oh, wait that's a horse, sorry. You need STABLE |
| 10:13.56 | Savage-S | lol |
| 10:14.08 | Zeeek | cheap shot, but what the heck |
| 10:14.15 | Vercingetorix | Back -- OK - That's what I did. How do I check out stable then? I've been downloading the prepared tarballs |
| 10:14.53 | Zeeek | isn't the syntax given on digium.com ? |
| 10:15.13 | Jas_Williams | Vercingetorix: follow directions on http://www.asterisk.org/index.php?menu=download |
| 10:15.16 | Vercingetorix | The sysntax is there, but there is no explanation as to what the commands mean. |
| 10:15.35 | Zeeek | I never needed to know :) |
| 10:15.35 | Jas_Williams | cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds |
| 10:15.52 | Jas_Williams | -r v1-0checks out version 1 = stable |
| 10:16.08 | Vercingetorix | Thanks. |
| 10:16.16 | *** join/#asterisk tessier (~treed@222.253.79.246) |
| 10:19.58 | *** join/#asterisk brenda (~nnnnn@c-67-182-205-227.hsd1.ut.comcast.net) |
| 10:20.13 | *** join/#asterisk gonzo- (~gonzo@lesnik.portaone.com) |
| 10:24.29 | dtwilson | having a wee first time look into sockets programming - thinking of using it within asterisk for a pseudo |
| 10:24.29 | dtwilson | <PROTECTED> |
| 10:24.29 | dtwilson | <PROTECTED> |
| 10:25.21 | *** join/#asterisk malabar (~malabar@164.80-202-124.nextgentel.com) |
| 10:27.04 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
| 10:45.20 | *** join/#asterisk cmk (~cmk_@p54A3CFAE.dip.t-dialin.net) |
| 10:46.10 | syle | yes, why redo whats already done |
| 10:46.27 | syle | so you gonna use a socket to accept then do all the bridging code to? |
| 10:47.37 | *** join/#asterisk _omer (dfsdf@202.147.174.176) |
| 10:49.26 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 10:55.06 | syle | if you do it my only pointer is use pthreads for each incomming call |
| 11:05.19 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 11:06.46 | vpp | docs ? |
| 11:08.22 | Zeeek | vpp what docs? |
| 11:09.01 | vpp | i was looking for the trigger that points to where the asterisk docs are |
| 11:09.07 | vpp | newbie pming hehe |
| 11:09.12 | Zeeek | ~docs |
| 11:09.13 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 11:09.16 | Zeeek | Starter tutorial: |
| 11:09.16 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 11:09.16 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 11:09.16 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 11:09.16 | Zeeek | THE reference of the moment: |
| 11:09.16 | Zeeek | http://www.asteriskdocs.org |
| 11:09.18 | vpp | :) |
| 11:09.56 | RoyK | ~lart Zeeek |
| 11:10.08 | Zeeek | how the farfon? |
| 11:10.16 | RoyK | dusty |
| 11:10.26 | Zeeek | at least my office door stays open now |
| 11:10.34 | *** join/#asterisk gambolputty (~gambolput@cblmdm69-45-216-83.buckeye-express.com) |
| 11:10.37 | RoyK | :) |
| 11:10.39 | Zeeek | nice rubber feet |
| 11:11.42 | Zeeek | hungry |
| 11:12.08 | vpp | anyone use a sangoma T1 card? |
| 11:12.29 | RoyK | E1 |
| 11:12.31 | RoyK | yes |
| 11:12.33 | RoyK | same thing |
| 11:13.11 | vpp | i'm finding that when i reboot it and i try to call out of it |
| 11:13.15 | vpp | it says congested |
| 11:13.20 | vpp | but if i do reload twice |
| 11:13.23 | vpp | its ok |
| 11:13.39 | vpp | is the card failing to negotiate with the other side? |
| 11:13.51 | vpp | ungracefull shutdown when asterisk is rebooted? |
| 11:13.56 | vpp | had any problems like that? |
| 11:16.36 | *** join/#asterisk Gunnar (~gunnar@99.82-134-107.bkkb.no) |
| 11:23.19 | *** join/#asterisk CdtDelta (~CdtDelta@dsl081-225-161.chi1.dsl.speakeasy.net) |
| 11:23.58 | RoyK | zoa: ping |
| 11:24.03 | RoyK | ~seen zoa |
| 11:24.05 | jbot | zoa is currently on #asterisk. Has said a total of 7 messages. Is idling for 1h 31m 57s |
| 11:24.12 | oej | zoa sleeping... |
| 11:24.23 | RoyK | kk |
| 11:24.45 | RoyK | oej: kan du anbefale noe godt hotell med wlan til VON? |
| 11:25.04 | oej | Vet inte riktigt vilka som har wlan... |
| 11:25.14 | oej | Mellanpris, lågpris, högpris? |
| 11:25.49 | RoyK | tja. sjefen sa <= 1500 per natt, så det kan vel bli noe bra, vil jeg tro...... gjerne billigere |
| 11:27.03 | RoyK | SEK 1700 per natt er vel ganske høyt |
| 11:27.11 | oej | Any hotel downtown Stockholm. We're running Meet Asterisk at Continental, check www.astricon.net/meetasterisk for info. It's by the train station that has the commuter train to the conference |
| 11:27.44 | RoyK | how far from the city center is the conference? |
| 11:27.56 | *** join/#asterisk Nix (~Nix@81.214.255.57) |
| 11:29.41 | RoyK | oej: meet asterisk doesn't really seem that interesting. I beleive I know what asterisk is :) |
| 11:30.04 | oej | You need to register for astricon |
| 11:31.14 | oej | ...or send me a talk proposal! |
| 11:32.19 | Romik | how to set callerid of zapata channels? |
| 11:32.29 | zoa | oej, we have a new version of the sip jb again |
| 11:32.35 | zoa | solves some more issues we found |
| 11:32.39 | oej | zoa: You're alive! |
| 11:34.04 | *** join/#asterisk prh (~paul@212.13.203.80) |
| 11:34.45 | RoyK | oej: astericon? |
| 11:34.52 | RoyK | oej: I was talking about the VON |
| 11:34.58 | oej | RoyK: I moved to Astricon :-) |
| 11:35.07 | *** join/#asterisk darby_t (~tom@host-ip226-209.crowley.pl) |
| 11:35.15 | RoyK | you're not going to the VON? |
| 11:35.39 | *** part/#asterisk darby_t (~tom@host-ip226-209.crowley.pl) |
| 11:36.16 | Nix | eoj. I may turn up to Astricon |
| 11:36.48 | Savage-S | hello, does anyone has experience on asterisk with BRI, with sub-adressing on the D-channel? |
| 11:39.48 | Savage-S | Asterisk sees the sub-adressing coming in, but it does not pass it through from external to internal channels and vice versa |
| 11:39.48 | oej | RoyK: Yes, I'm speaking at VoN |
| 11:39.49 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 11:39.49 | Nix | RoyK: you are in .ie right? |
| 11:39.49 | dtwilson | RoyK: are you in .ie? so am I :) |
| 11:40.29 | RoyK | I'm in .no |
| 11:41.00 | RoyK | zoa: another new one? |
| 11:41.12 | dtwilson | ahhh - ok, was surprised to see if there were others here from .ie |
| 11:41.15 | zoa | yes |
| 11:41.16 | zoa | another one |
| 11:41.41 | RoyK | can I have that url again? |
| 11:41.49 | RoyK | reinstalled the powerbook with a 7.2k spin drive....... |
| 11:41.52 | RoyK | bookmarks gone |
| 11:42.03 | Nix | I need a DID in .ie |
| 11:42.17 | Nix | anyone have any recomendations? its for a customer |
| 11:42.39 | Nix | preferably in dublin |
| 11:45.07 | *** join/#asterisk ilium007 (~brantwint@220-253-92-177.QLD.netspace.net.au) |
| 11:45.22 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 11:45.26 | lehel | hello |
| 11:45.34 | lehel | tzafrir: are you here? |
| 11:45.37 | Romik | Nix: I need in germany, japan, italy, france, spain do you know where i can order such DIDs? |
| 11:47.26 | Nix | in germany it is still illegal afaik |
| 11:47.35 | Nix | japan maybe |
| 11:47.38 | Nix | the others no |
| 11:47.40 | Nix | sorry :-( |
| 11:47.56 | Romik | Nix: iconnecthere offer them, but only need their device - native work only with USA |
| 11:48.13 | Romik | nix: japan which provider? |
| 11:48.22 | Nix | I have a japanese partner |
| 11:48.25 | Delvar | Nix: voiptalk.org do sip UK DDI's including dublin |
| 11:48.27 | *** join/#asterisk Mavvie (edwin@edwin.adsl.barnet.com.au) |
| 11:48.42 | Nix | Delvar: I couldnt find dublin DIDs in voiptalk :-( |
| 11:48.51 | Delvar | o_0 |
| 11:49.28 | Nix | are you sure? |
| 11:50.25 | Nix | bbl |
| 11:51.09 | Delvar | lol my mistake |
| 11:52.07 | Zeeek | Romik look at the asterisk-biz mailing list - this is discussed daily |
| 11:58.51 | Romik | zeeek: where i check this list? |
| 11:59.15 | Romik | zeee: found it |
| 11:59.36 | lehel | line 0: Unable to open master device '/dev/zap/ctl' |
| 12:00.11 | lehel | ?? |
| 12:00.13 | lehel | pls |
| 12:00.23 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
| 12:01.10 | eper-werk | I hope telappliant/voiptalk sent my card out soon, there support is not very responsive |
| 12:01.56 | *** join/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org) |
| 12:01.58 | *** join/#asterisk Mother__ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
| 12:02.06 | Mother__ | multiple greetings |
| 12:02.13 | Mavvie | lehel: do some basic trouble shooting. does the file exist? |
| 12:03.05 | lehel | Mavvie: ?? on my card aren't the lights on! :( |
| 12:03.16 | lehel | i have problems with zaptel configuration |
| 12:03.23 | Mavvie | lehel: still, do some basic troubble shooting: does the file exist? |
| 12:03.25 | lehel | PPP... ??? |
| 12:03.26 | Mother__ | quick question: in zapata.conf I have a zap channel as pickupgroup=1, then a SIP channel also has pickupgroup=1 in sip.conf, but I get "chan_sip.c:7802 handle_request: Nothing to pick up" |
| 12:03.57 | lehel | Mavvie: i can't understood you! what file?? |
| 12:04.11 | Mother__ | I also have group=1 and callgroup=1 in zapata.conf for this channel |
| 12:04.14 | Mavvie | lehel: the one mentioned in the line you pasted. |
| 12:04.30 | lehel | exists!.. |
| 12:04.46 | Mavvie | lehel: go on... |
| 12:05.03 | lehel | ?go on? |
| 12:05.03 | lehel | cannot open the file |
| 12:05.21 | Mother__ | any ideas?? |
| 12:05.31 | Mavvie | lehel: now it's up to you to figure out when you can't open a file. |
| 12:05.51 | *** join/#asterisk O-Zone_ (~O-Zone@moloch.asb.unisi.it) |
| 12:05.53 | Mavvie | s/when/why |
| 12:05.57 | O-Zone_ | hi all |
| 12:05.58 | *** join/#asterisk cursor (~kevin@andromeda.office.cursor.biz) |
| 12:06.09 | cursor | hello all |
| 12:06.28 | Mavvie | I guess that was too difficult |
| 12:07.17 | O-Zone_ | i need some help to asterisk |
| 12:07.29 | Mother__ | lol |
| 12:07.40 | *** join/#asterisk Renfield (~Renfield@24-54-42-81.lndnnh.adelphia.net) |
| 12:07.47 | cursor | What sort of help do you need? |
| 12:08.04 | Mavvie | O-Zone_: asterisk is not a verb. |
| 12:08.08 | Mother__ | hehe |
| 12:08.10 | O-Zone_ | ok |
| 12:08.14 | O-Zone_ | i've installed asterisk |
| 12:08.16 | O-Zone_ | from CVS |
| 12:08.20 | O-Zone_ | with AMP |
| 12:08.33 | O-Zone_ | i wish to use it as a PBX for SIP phones |
| 12:08.38 | O-Zone_ | into an office |
| 12:08.51 | O-Zone_ | how i can add SIP users ? |
| 12:08.56 | O-Zone_ | i've added peers to sip.conf |
| 12:09.03 | O-Zone_ | and now asterisk accept registration |
| 12:09.11 | O-Zone_ | but deny calls between peers |
| 12:09.51 | cursor | You need sip users and peers in sip.conf |
| 12:10.16 | cursor | or, if they are just phones then you can use "friend" |
| 12:10.47 | cursor | friend is evil for service provider links, but is ok for phones |
| 12:10.48 | Mother__ | grrrr I just cannot get *8# to work!!! |
| 12:11.19 | Mother__ | it's bound to be something silly, but I have added group pickupgroup and callgroup almost everywhere now |
| 12:11.29 | O-Zone_ | cursor: i try following the instruction to add friends from mysql |
| 12:11.32 | O-Zone_ | but don't work |
| 12:11.33 | *** join/#asterisk stormfr (~StorM@82.66.251.138) |
| 12:11.42 | cursor | this should be all you need: |
| 12:11.42 | cursor | callgroup = 1 |
| 12:11.42 | cursor | pickupgroup = 1 |
| 12:11.46 | cursor | in sip.conf |
| 12:11.48 | cursor | and wherever |
| 12:11.55 | cursor | depending upon the phones :-) |
| 12:12.08 | O-Zone_ | an example of my sip.conf peer |
| 12:12.10 | O-Zone_ | [duccio] |
| 12:12.10 | O-Zone_ | secret=duccio |
| 12:12.10 | O-Zone_ | type=friend |
| 12:12.10 | O-Zone_ | host=dynamic |
| 12:12.10 | O-Zone_ | context=sipexts |
| 12:12.11 | O-Zone_ | mailbox=9903 |
| 12:12.13 | O-Zone_ | callerid="duccio@asb.unisi.it" <9903> |
| 12:12.15 | O-Zone_ | dmtfmode=rfc2833 |
| 12:12.17 | O-Zone_ | canreinvite=no |
| 12:12.22 | Savage-S | hello, does anyone has experience on asterisk with BRI, with sub-adressing on the D-channel? |
| 12:12.23 | O-Zone_ | 9903 is the extension i've created with AMP |
| 12:12.23 | Savage-S | Asterisk sees the sub-adressing coming in, but it does not pass it through from external to internal channels and vice versa |
| 12:12.24 | Mother__ | cursor: Cisco 7912s |
| 12:12.51 | cursor | O-zone: username = duccio |
| 12:13.46 | O-Zone_ | ok |
| 12:14.23 | O-Zone_ | i've added callgroup = 1 and pickupgroup = 1 to each peer |
| 12:14.40 | cursor | :-) |
| 12:14.46 | Mother__ | this are the (L)user phones |
| 12:14.49 | Mother__ | s/this/these |
| 12:15.26 | O-Zone_ | cursor: ufff...403 - not found |
| 12:15.51 | cursor | OZ: Are the phones on the same network as the Asterisk box? |
| 12:15.56 | *** part/#asterisk Renfield (~Renfield@24-54-42-81.lndnnh.adelphia.net) |
| 12:16.00 | cursor | I.e. Are there any NAT/firewall issues? |
| 12:16.20 | O-Zone_ | Looking for fabrizio in sipexts |
| 12:16.20 | O-Zone_ | Reliably Transmitting (NAT): |
| 12:16.20 | O-Zone_ | SIP/2.0 404 Not Found |
| 12:16.20 | O-Zone_ | Via: SIP/2.0/UDP 192.167.125.9:5064;branch=z9hG4bK55299E33;received=192.167.125.9;rport=5064 |
| 12:16.20 | O-Zone_ | From: "michele" <sip:michele@asb.unisi.it>;tag=77FCEB38 |
| 12:16.21 | O-Zone_ | To: <sip:fabrizio@asb.unisi.it>;tag=as0dbe1be9 |
| 12:16.23 | O-Zone_ | Call-ID: 1380090725@192.167.125.9 |
| 12:16.25 | O-Zone_ | CSeq: 1902 INVITE |
| 12:16.27 | O-Zone_ | User-Agent: Asterisk PBX |
| 12:16.29 | O-Zone_ | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER |
| 12:16.31 | O-Zone_ | Contact: <sip:fabrizio@192.167.125.9> |
| 12:16.33 | O-Zone_ | Content-Length: 0 |
| 12:16.36 | O-Zone_ | cursor: yes |
| 12:16.38 | O-Zone_ | there's a firewall |
| 12:16.56 | O-Zone_ | but don't blocki anything between us |
| 12:17.01 | RoyK | ~pastebin |
| 12:17.01 | cursor | ok |
| 12:17.07 | Mother__ | hah it verks!!! |
| 12:17.10 | O-Zone_ | sorry |
| 12:17.11 | O-Zone_ | :-p |
| 12:17.23 | cursor | OZ: by the way - change your password when you've finished :-) |
| 12:17.34 | cursor | I have your IP, username and password now - you pasted them all |
| 12:17.34 | Mother__ | thanks all |
| 12:17.49 | Mother__ | cya cursor, thanks |
| 12:17.54 | cursor | no probs |
| 12:17.57 | O-Zone_ | cursor: you can't do anything...sorry :-D |
| 12:18.01 | cursor | :-) |
| 12:18.09 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 12:18.20 | cursor | I wouldn't anyway, but I can't speak for everyone here :-) |
| 12:18.23 | O-Zone_ | cursor: i'm behind a firewalled network |
| 12:18.29 | cursor | Good |
| 12:18.31 | O-Zone_ | there's no problems |
| 12:18.33 | O-Zone_ | :-D |
| 12:18.41 | O-Zone_ | there are dummy accounts |
| 12:19.07 | O-Zone_ | i only want to make asterisk work :-D |
| 12:19.09 | O-Zone_ | ideas ? |
| 12:19.15 | cursor | do your phones show up if you "sip show peers" |
| 12:19.20 | cursor | and "sip show users" |
| 12:19.38 | cursor | and "sip show registry" |
| 12:19.39 | cursor | etc. |
| 12:19.58 | *** join/#asterisk durex (~ironman@200.101.109.138) |
| 12:20.52 | t0p | what DNS records do I have to add to allow SIP related queries |
| 12:21.19 | cursor | nothing |
| 12:21.19 | O-Zone_ | moloch*CLI> sip show peers |
| 12:21.19 | O-Zone_ | Name/username Host Dyn Nat ACL Mask Port Status |
| 12:21.19 | O-Zone_ | duccio/duccio (Unspecified) D N 255.255.255.255 0 Unmonitored |
| 12:21.19 | O-Zone_ | fabrizio/fabriz 192.167.125.12 D N 255.255.255.255 5060 Unmonitored |
| 12:21.19 | O-Zone_ | michele/michele 192.167.125.9 D N 255.255.255.255 5064 Unmonitored |
| 12:21.19 | O-Zone_ | moloch*CLI> |
| 12:21.25 | O-Zone_ | seems yes |
| 12:21.26 | cursor | unless you want SRV records too |
| 12:21.35 | *** join/#asterisk fcbouan (~Franky@stardust.noc.frontier.fr) |
| 12:22.02 | O-Zone_ | sip show registry is empty |
| 12:22.06 | O-Zone_ | what it mean ? |
| 12:22.16 | cursor | nothing is registered |
| 12:22.25 | O-Zone_ | cursor: so ? |
| 12:22.29 | O-Zone_ | if i open kphone |
| 12:22.33 | O-Zone_ | it say that i'm registered ! |
| 12:22.37 | cursor | check your phones to make sure they are trying to register with Asterisk |
| 12:22.43 | O-Zone_ | ..after configuration |
| 12:22.48 | t0p | cursor: so, if someone calls me at me@mydomain.com as I understand there will be a query to one of my DNS servers |
| 12:23.08 | cursor | t0p if you have an Asterisk box on mydomain.com then that's fine |
| 12:23.29 | cursor | if your Asterisk box is on voip.mydomain.com then you can use that as a SIP address |
| 12:23.30 | t0p | cursor: i see |
| 12:23.38 | fcbouan | hi , can someone help me ? i need use ast_mutex_trylock() instead of ast_mutex_lock() for mysql query in c module ? |
| 12:23.47 | cursor | and you can also set up SRV records to point mydomain.com -> voip.mydomain.com |
| 12:23.54 | cursor | as a shortcut |
| 12:24.06 | t0p | cursor: so, it goes to whatever IP it resolves from mydomain.com |
| 12:24.08 | O-Zone_ | mmmm |
| 12:24.11 | cursor | right |
| 12:24.23 | O-Zone_ | a pastebin site ? |
| 12:24.26 | cursor | unless mydomain.com has a SRV record pointing SIP elsewhere |
| 12:24.36 | cursor | pastebin.ca |
| 12:24.45 | vaewynAFK | patebin.ca |
| 12:24.47 | t0p | cursor: okay, I understand now |
| 12:24.55 | *** join/#asterisk clive- (~pirch@rndf-146-52-213.telkomadsl.co.za) |
| 12:24.56 | vaewyn | pastebin.ca even :P |
| 12:25.28 | cursor | patebin is used to throw away meat spreads |
| 12:25.33 | clive- | anyone here familar with chan_capi, I am trying to increase the gain, without success |
| 12:26.17 | O-Zone_ | http://pastebin.ca/11659 << all REGISTER flow |
| 12:26.21 | O-Zone_ | what0's wrong !??!? |
| 12:26.32 | O-Zone_ | my colleaugue Fabrizio say that receives ACK from me ! |
| 12:26.35 | cursor | I don't use capi, but I understand it's a digital line, so it probably doesn't have gains, as sucj |
| 12:26.36 | cursor | such |
| 12:27.34 | *** join/#asterisk michal_ (~michal@lts-cna.fpf.slu.cz) |
| 12:27.34 | clive- | cursor there is a gain setting, but it doesnt seem to have an effect |
| 12:27.59 | cursor | ok - I don't use CAPI, so I don't really know |
| 12:28.18 | cursor | No ISDN here |
| 12:28.31 | cursor | SIP/IAX links only here |
| 12:29.34 | cursor | O-Z perhaps you need nat=yes in your friend block |
| 12:31.41 | O-Zone_ | cursor: i've added it |
| 12:31.43 | O-Zone_ | :-P |
| 12:31.50 | O-Zone_ | now it say: Service not available |
| 12:33.04 | fcbouan | is there someone who can help me for ast_mutex_lock usage ? |
| 12:33.08 | *** join/#asterisk gbdrbob (drbob@alltalk.demon.co.uk) |
| 12:33.14 | gbdrbob | Hello |
| 12:33.17 | cursor | hello |
| 12:33.23 | fcbouan | hi |
| 12:33.26 | gbdrbob | I've found a minor bug |
| 12:33.26 | O-Zone_ | SIP/2.0 503 Service Unavailable |
| 12:33.34 | gbdrbob | In the monitor app |
| 12:34.20 | cursor | in CVS v1-0 or HEAD (or a package)? |
| 12:34.23 | sudhir492 | what is the caller id variable in the channel? |
| 12:34.33 | cursor | ${CALLERID} |
| 12:34.35 | cursor | ${CALLERIDNAME} |
| 12:34.37 | gbdrbob | in stable |
| 12:34.37 | cursor | ${CALLERIDNUM} |
| 12:34.56 | gbdrbob | if the monitor filename has a space in it them the automated mix/xleanup of the "m" option doesn't work |
| 12:35.00 | sudhir492 | for some readon $CALLERID is blank ! |
| 12:35.08 | cursor | I use NAME/NUM |
| 12:35.40 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 12:36.07 | sudhir492 | thanks |
| 12:36.14 | sudhir492 | CALLERIDNUM is ok |
| 12:36.32 | gbdrbob | I put the callerid in the monitor filename and when it is withheld * substitues "CID withheld" the extra space stops rm working. |
| 12:36.38 | Mavvie | CALLERIDIUM sounds like some kind of element. |
| 12:37.36 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 12:37.59 | gbdrbob | the monitor app should escape space in the filename or put the whole filename in quotes when calling rm |
| 12:38.10 | cursor | The monitor app shouldn't call rm |
| 12:38.17 | cursor | it should call unlink() |
| 12:38.28 | durex | folks.... |
| 12:38.32 | cursor | Just modify this line to add quotes: |
| 12:38.32 | cursor | snprintf(tmp2,sizeof(tmp2), "( %s& rm -f \"%s\"/%s-* ) &",tmp, dir ,name); /* remove legs when done mixing */ |
| 12:38.41 | cursor | move the existing end quote |
| 12:38.52 | cursor | err |
| 12:38.56 | durex | does somebody knows how to identify a 'realm' in a sip register ? |
| 12:39.21 | cursor | no |
| 12:39.29 | cursor | moving the end quote won't work :-) |
| 12:39.48 | marlowe | is simpletelcom still i nbusiness? |
| 12:39.55 | marlowe | they have been unreachable for me for about a month now |
| 12:39.55 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 12:40.09 | cursor | Never heard of simpletelcom |
| 12:40.43 | marlowe | I dunno - I dont use them - They were backup like #5 but still.. The web site is up and all, I can login. They're just unreachable - No replies to emails. |
| 12:42.28 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 12:44.07 | cursor | gbdrbob: I could give you a quick and dirty fix to try |
| 12:44.19 | t0p | cursor: would you know the pattern to add SRV record in bind? |
| 12:44.34 | cursor | _sip._udp IN SRV 10 0 5060 voip |
| 12:44.40 | cursor | where voip is in voip.yourdomain.com |
| 12:44.59 | t0p | cursor: thanks |
| 12:45.00 | O-Zone_ | what is they ? |
| 12:45.01 | O-Zone_ | SIP/2.0 200 OK |
| 12:45.02 | O-Zone_ | Via: SIP/2.0/UDP 192.167.125.9:5060;rport;branch=z9hG4bK392e695e |
| 12:45.02 | O-Zone_ | From: "Unknown" <sip:Unknown@192.167.125.9>;tag=as43a3b05d |
| 12:45.02 | O-Zone_ | CSeq: 102 NOTIFY |
| 12:45.02 | O-Zone_ | Call-ID: 34e9682d0abf934432d005735d450bd3@192.167.125.9 |
| 12:45.03 | O-Zone_ | To: <sip:michele@192.167.125.9:5062;transport=udp>;tag=4FC19A06 |
| 12:45.05 | O-Zone_ | Content-Length: 0 |
| 12:45.08 | O-Zone_ | User-Agent: kphone/4.0.5 |
| 12:45.09 | O-Zone_ | Contact: "michele" <sip:michele@192.167.125.9:5062;transport=udp> |
| 12:45.18 | marlowe | O: www.pastebin.ca next time |
| 12:45.38 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-209-247.dsl.scarlet.be) |
| 12:47.37 | O-Zone_ | why i can't register to my asterisk ? |
| 12:48.41 | cursor | michele@192.167.125.9:5062 <-- weird, but maybe ok |
| 12:48.44 | cursor | 5062 ? |
| 12:50.07 | Mavvie | cursor: it's okay |
| 12:50.22 | gbdrbob | cursor: go for it :-) |
| 12:50.43 | cursor | "res/res_monitor.c" |
| 12:50.48 | cursor | approx. line 251 |
| 12:50.50 | O-Zone_ | cursor: asterisk is in the same machine of michele's sip client |
| 12:50.56 | cursor | Change this: |
| 12:50.56 | cursor | "(tmp2,sizeof(tmp2), "( %s& rm -f \"%s\"/%s-* ) &" |
| 12:50.58 | cursor | to this: |
| 12:51.02 | cursor | "( %s& find \"%s\" -name \"%s-*\" -maxdepth 1 -exec /bin/rm -f \"{}\" \\; ) &" |
| 12:51.08 | cursor | I said it was dirty :-) |
| 12:52.02 | cursor | and I haven't tested it, so beware :-) |
| 12:52.38 | cursor | oops - don't change all of that |
| 12:52.49 | cursor | just the part that starts with "( %s& |
| 12:52.49 | gbdrbob | lol |
| 12:53.41 | cursor | Change this much of it: |
| 12:53.41 | cursor | "( %s& rm -f \"%s\"/%s-* ) &" |
| 12:53.50 | gbdrbob | ok |
| 12:54.38 | RoyK | does anyone have an overview over asterisk memory requirements? |
| 12:54.47 | RoyK | my main sip gateway uses 1GB RAM |
| 12:54.52 | cursor | yes - Asterisk requires memory |
| 12:55.20 | *** join/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net) |
| 12:58.02 | cursor | (RoyK) Our Asterisk server has 512MB and sits comfortably in there |
| 12:58.06 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 12:58.14 | cursor | I suppose it depends upon your usage and the number of phones/links etc. |
| 12:58.28 | clive- | patrick, hi,,,, |
| 13:01.00 | RoyK | cursor: what traffic? |
| 13:01.07 | cursor | not a lot |
| 13:01.19 | cursor | run "top" and see what Asterisk is using |
| 13:01.33 | cursor | Press "M" to sort by memory usage |
| 13:02.37 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 13:03.09 | RoyK | cursor: I know :P |
| 13:03.17 | cursor | :-) |
| 13:03.18 | RoyK | cursor: and asterisk uses a gig or so |
| 13:03.26 | cursor | <PROTECTED> |
| 13:03.26 | cursor | 10199 asterisk -11 0 29976 6576 3748 S 0.0 1.0 0:00.35 asterisk |
| 13:03.33 | cursor | Ours looks like that |
| 13:03.36 | cursor | several of those lines |
| 13:03.41 | fcbouan | is there someone who can help me for ast_mutex_lock usage ? |
| 13:03.46 | RoyK | 13766 ? S< 0:12 0 605 943582 853508 41.1 \_ asterisk -vvvg -c |
| 13:03.50 | cursor | 12 of them |
| 13:04.03 | RoyK | 54 |
| 13:04.09 | RoyK | not too much trafiic now..... |
| 13:04.12 | RoyK | on that box |
| 13:04.49 | cursor | fcbouan: There should be plenty of examples in the code |
| 13:05.11 | fcbouan | i got lot of deadlock |
| 13:05.14 | fcbouan | with mysq ltt |
| 13:05.29 | fcbouan | with realtime |
| 13:05.43 | RoyK | what sort of deadlocks? |
| 13:05.44 | fcbouan | even if i remove all lock/unlock in my app |
| 13:06.04 | fcbouan | deadlock on mysql query |
| 13:06.34 | fcbouan | is module use separed ressource on mutex lock or can interact ? |
| 13:06.48 | fcbouan | realtime lock interact with my custom mod |
| 13:07.04 | RoyK | does anyone know why I can't "sip show peer xxx" with realtime? |
| 13:07.05 | RoyK | that sucks |
| 13:10.47 | cursor | brb... |
| 13:11.58 | newl | RoyK: you can, use the load option. |
| 13:12.19 | newl | see also: sip show peer<return> :) |
| 13:12.40 | O-Zone_ | why my sip phones don't register on asterisk !?!?? |
| 13:13.05 | newl | Because it may be improperly configured? |
| 13:13.27 | O-Zone_ | but my kphone say to me that i'm registered ! |
| 13:15.00 | clive- | any capi users around? |
| 13:15.51 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 13:20.08 | t0p | Hi, is this exten => _1644,1,Dial(SIP/myfriend@otherdomain.com) correct? if my * has a real IP and the /etc/resolv.conf points to working name servers? |
| 13:20.26 | t0p | i'm on FC3 |
| 13:22.25 | cursor | t0p: looks ok |
| 13:22.40 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 13:22.40 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 13:23.06 | O-Zone_ | ufff |
| 13:23.12 | O-Zone_ | between sip phone connected to asterisk server |
| 13:23.13 | t0p | cursor: :-) , it's my first day getting * to run |
| 13:23.23 | O-Zone_ | can i call using username or i need to allocate numbers ? |
| 13:23.48 | cursor | numbers and usernames are interchangeable |
| 13:23.52 | O-Zone_ | ok |
| 13:23.54 | cursor | it depends upon your setup |
| 13:24.03 | cursor | foo@bar.com |
| 13:24.07 | cursor | foo can be a number |
| 13:24.10 | cursor | if defined |
| 13:24.13 | O-Zone_ | my setup don't work :-( |
| 13:25.08 | *** join/#asterisk iq (~iq@63-230-44-112.omah.qwest.net) |
| 13:25.09 | t0p | cursor: I tried name-to-number mapping in extensions.conf but it didn't seem to work |
| 13:25.22 | *** part/#asterisk clive- (~pirch@rndf-146-52-213.telkomadsl.co.za) |
| 13:26.02 | cursor | exten => foo,... |
| 13:26.44 | cursor | I have things like this: exten => echo,1,Goto(2009,1) |
| 13:26.56 | cursor | so that maps echo to 2009 |
| 13:26.59 | cursor | extension 2009 is a local echo test |
| 13:27.09 | t0p | cursor: ok, i get it working now |
| 13:27.14 | t0p | cursor: thanks |
| 13:27.26 | *** part/#asterisk n4y (~frodo7@host-ip226-209.crowley.pl) |
| 13:27.36 | O-Zone_ | mmm |
| 13:27.48 | O-Zone_ | users are extension ? |
| 13:27.55 | O-Zone_ | i'm going crazy |
| 13:28.18 | cursor | "user" defines an incoming connection |
| 13:28.24 | cursor | "peer" defines an outgoing connection |
| 13:28.26 | O-Zone_ | yes |
| 13:28.39 | O-Zone_ | but Asterisk can handle REGISTER for SIP users ? |
| 13:28.44 | newl | "friend" defines someone you go out drinking with. |
| 13:28.47 | O-Zone_ | why uit don't wok |
| 13:28.47 | zoa | yes it can |
| 13:28.59 | O-Zone_ | there's some docs hot to do it |
| 13:29.11 | cursor | It all depends upon who's buying |
| 13:29.22 | newl | hehe |
| 13:29.37 | O-Zone_ | ? |
| 13:29.41 | RoyK | zoa: can I have the download url to the jb again, please? |
| 13:29.53 | zoa | the newest one is not online yet |
| 13:31.15 | cursor | remove legs when done mixing <-- ouch |
| 13:31.19 | *** join/#asterisk pussfeller (~todd@t1-rtc-woodlawn.rtcol.com) |
| 13:31.53 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 13:32.08 | cjk | hi,does anyone know a tool like sipsak for iax? |
| 13:32.50 | cursor | If there is one then it'll be in the Asterisk CVS |
| 13:33.00 | Sato1 | whats sipsak? |
| 13:33.03 | *** join/#asterisk ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 13:33.23 | cursor | IAX is proprietory - SIP is standard, so more people work with it |
| 13:33.53 | cursor | proprietary, even |
| 13:33.55 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 13:34.03 | Sato1 | hi ariel_, remember my problem with my FWD? its a FWD problem |
| 13:34.23 | ariel_ | Sato1, great to hear it's not on your end. |
| 13:34.30 | ariel_ | morning everyone. |
| 13:34.34 | Sato1 | still a problem |
| 13:34.45 | Sato1 | got another account and now its working fine |
| 13:35.00 | cursor | ariel: Afternoon |
| 13:35.02 | RoyK | zoa: ok |
| 13:35.16 | ariel_ | cursor, good afternoon to you. |
| 13:35.20 | RoyK | zoa: do you know when? |
| 13:35.33 | zoa | cjk, astertest does it a little |
| 13:35.37 | ariel_ | Just want to let everyone know it's a good Friday 13th.... |
| 13:35.43 | zoa | as well as massregister.tar.gz |
| 13:36.46 | Sato1 | thats the only thing i hate about my birthday, some times next day is Friday 13th |
| 13:37.15 | *** join/#asterisk jackthe (~jesse@d594f03e.ftth.concepts.nl) |
| 13:37.34 | ManxPower | Don't download http://www.fnords.org/~eric/asterisk/wffs.tar.gz It contains all sort of useful AGI scripts and dialplan examples and config examples and ITU source code for G729 and G723.1 |
| 13:37.39 | Nuxi | We're updating the firmware on all of our motorola vt1000s today just to test fate. |
| 13:38.06 | RoyK | ManxPower: thanks. I won't |
| 13:38.24 | ManxPower | RoyK: 8-) |
| 13:38.39 | cursor | Sato1: Don't worry - Friday 13 was only really unlucky in 1307 |
| 13:39.18 | iCEBrkr | ha |
| 13:39.18 | Nuxi | ManxPower, I'm having trouble getting some of the G729 AGI scripts you just mentioned ... |
| 13:39.29 | *** join/#asterisk OzJames79 (~opera@CPE20320889-1842-1.gex.ncable.net.au) |
| 13:39.46 | ManxPower | Nuxi: I didn't mention any G729 AGI scripts. |
| 13:40.12 | Nuxi | You mean there are no G729 AGI scripts at http://www.fnords.org/~eric/asterisk/wffs.tar.gz |
| 13:40.22 | ManxPower | http://www.fnords.org/~eric/asterisk/wffs.tar.gz |
| 13:40.35 | cursor | ManxPower: I don't think the ITU allow you to distribute their src, but I don't really care either way :-) |
| 13:40.48 | ManxPower | Nuxi: There are no G729 AGI scripts there. There are AGI scripts and there is ITU G729 source code, but they are not the same thing. |
| 13:41.07 | ManxPower | cursor: *shrug* They let me download it for free so they can't be THAT concerned about it.\ |
| 13:41.17 | cursor | :-) |
| 13:41.45 | cursor | I've deliberately not looked at it |
| 13:42.25 | cursor | From what I'm told, the ITU source is a bit of a mess anyway |
| 13:42.38 | blitzrage | morning all |
| 13:42.43 | Sato1 | whats the difference between that ITU g729 and the intel code for g729? |
| 13:42.52 | cursor | Arrest of Jaques de Molay, Grand Master of the Knights Templar, and 60 of his senior knights |
| 13:43.01 | ManxPower | cursor: Oh yes. It doesn't even compile out of the box. But I figure it's a good joke to let people download it and then know they will spend hours of wasted time trying to make it work. |
| 13:43.03 | blitzrage | ManxPower: I'm going to really start thinking you're a bot when you don't sleep :) |
| 13:43.03 | Nuxi | Sato1, TU != ntel |
| 13:43.05 | cursor | Friday 13 October 1307 |
| 13:43.15 | Sato1 | oh |
| 13:43.29 | ManxPower | blitzrage: I slept a total of 8 hours out of the paste 24 hours. |
| 13:43.36 | ManxPower | I just sleep at weird hours |
| 13:43.37 | blitzrage | ManxPower: yah, I imagine so |
| 13:43.52 | cursor | lol@manxpower |
| 13:43.56 | blitzrage | ManxPower: you were awake when I went to bed at 3:30, and now you're away when I wake up at 9:45am :) |
| 13:44.02 | Sato1 | so whats the difference? which one has better perfomance? ow.. which one its better? |
| 13:44.06 | ManxPower | 5pm - 9pm, 4am - 8am |
| 13:44.20 | blitzrage | oh, I can't split my sleeping like that :) |
| 13:44.25 | cursor | 10am - 10pm - 3 days later |
| 13:44.27 | blitzrage | wish I could |
| 13:44.50 | ManxPower | blitzrage: I can only do so if the splits are around specific times. I get REALLY tired 3pm - 6pm every day. |
| 13:45.15 | cursor | 2:45pm here now :-) |
| 13:45.21 | zoa | the itu one is better |
| 13:45.27 | zoa | it also works on non intel |
| 13:45.33 | ManxPower | Ugh. I need to call two places in Amsterdam today. |
| 13:45.40 | eper-werk | bit of a silly question however, does asterisk have anything like voice recorgnition or a "say the persons name" and it connects you to there extention? |
| 13:45.46 | Sato1 | the intel works actually on non intell procesors |
| 13:45.48 | ManxPower | eper-werk: no. |
| 13:45.57 | cursor | ManxPower you should keep that to yourself :-) |
| 13:46.07 | ManxPower | cursor: LOL! |
| 13:46.10 | ManxPower | cursor: HOTELS |
| 13:46.44 | ManxPower | "I want the readhead, number 4873. Can you ship via fedex? |
| 13:47.11 | cursor | :-) |
| 13:47.15 | *** join/#asterisk zamsler (~zamsler@c-67-175-210-62.hsd1.il.comcast.net) |
| 13:47.26 | cursor | It's only red hair under that lighting |
| 13:47.45 | Sato1 | zoa, it does not compile in non intel procesors, but if you compile it in a intel procesor and copy to an AMD, it works |
| 13:48.40 | sudhir492 | my asterisk occasionally crashes somewhere in pwlib :-( |
| 13:48.44 | *** join/#asterisk jterrero (~jterrero@mcse-irc.isys-networks.com) |
| 13:49.05 | Nuxi | eper-werk, you can do that with a little bit of agi and signal processing. |
| 13:49.45 | cursor | You'd do better with a directory list and extension numbers |
| 13:50.02 | Dishwasha | yeah |
| 13:50.02 | cjk | zoa: thanks btw |
| 13:50.19 | Dishwasha | wow, can't believe I stayed on irc idle all night, I normally get disconnected |
| 13:50.47 | Nuxi | Actually, it works quite well. I have about a dozen people that I can dial by saying their name. |
| 13:51.17 | blitzrage | cursor: you're fired! |
| 13:51.38 | Jas_Williams | 22 |
| 13:51.43 | zamsler | 23 |
| 13:51.43 | cursor | Dr. Michael Hfuhruhurr |
| 13:51.57 | cursor | Try getting voice recognition to find that |
| 13:52.09 | blitzrage | 24? |
| 13:52.11 | cursor | From "The Man with Two Brains" |
| 13:52.37 | Nuxi | eper-werk, the trick is to not use ASR packages meant for dictation. |
| 13:52.38 | cursor | Steve Martin classic |
| 13:53.00 | Nuxi | cursor, it will have no problem with Dr. Michael Hfuhruhurr. |
| 13:53.14 | cursor | :-) |
| 13:53.57 | cursor | It might get confused with Dr. Oppenheimer |
| 13:54.12 | sudhir492 | Hfuhruhurr :-) Is that a real name ? |
| 13:54.19 | cursor | It is in that film :-) |
| 13:54.35 | zamsler | sounds german |
| 13:54.46 | cursor | Sounds gibberish |
| 13:54.50 | zamsler | lol |
| 13:55.05 | cursor | Just try to say it |
| 13:55.48 | cursor | Ich spreche Deutsches wenig, aber nicht sehr gut. |
| 13:55.58 | *** join/#asterisk jskcr|lappy (~jskcr@jskcr.user) |
| 13:56.03 | RoyK | skjønte det.... |
| 13:56.13 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 13:56.42 | cursor | 9 :-) |
| 13:56.44 | *** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 13:56.55 | zamsler | 11 |
| 13:57.04 | RoyK | Nuxi: ya? |
| 13:57.06 | RoyK | JA! |
| 13:57.07 | RoyK | bitte |
| 13:58.11 | Nuxi | is JA the java version of ya? |
| 13:58.20 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 13:58.53 | RoyK | :) |
| 13:58.57 | cursor | Rho srapped mg kegtops awound? |
| 13:59.27 | ManxPower | Apparently the 468* "factory reset" on the Polycom phones does NOT clear out the "directory" on the phone. |
| 13:59.41 | ManxPower | You know the directory that defaults to silent ring for people in the directory |
| 13:59.57 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 13:59.57 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 14:02.25 | *** part/#asterisk OzJames79 (~opera@CPE20320889-1842-1.gex.ncable.net.au) |
| 14:03.12 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 14:03.12 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 14:03.53 | bkw_ | haha |
| 14:04.06 | cursor | :-) |
| 14:04.24 | zamsler | cursor.. Long as u don't flash.. u should be ok. |
| 14:04.26 | zoa | hey ho brian |
| 14:04.27 | zamsler | ;) |
| 14:04.27 | *** join/#asterisk jets (~brian@guardian.pmt.org) |
| 14:04.28 | Nuxi | does * run un hubble? |
| 14:04.47 | zoa | brian, do you remember the type of wifi phone file was using at von ? |
| 14:04.59 | cursor | No - hubble just looks at the * |
| 14:05.25 | *** join/#asterisk _-Jon-_ (jon@hs-wolfedale-216048166226.3web.net) |
| 14:05.58 | cursor | Why use WiFi phones when you can use DECT? |
| 14:06.20 | RoyK | good question |
| 14:06.33 | cursor | Saves nuking your brain |
| 14:06.54 | cursor | I wouldn't want a WiFi transmitter next to my head |
| 14:07.17 | Sato1 | headche |
| 14:07.42 | cursor | Gesundheit! |
| 14:08.13 | Sato1 | well, depends on the power, but 300mw... enough to give a good headche for a little while |
| 14:08.15 | Zeeek | ManxPower why are you dreading calling Amsterdam? They all speak English |
| 14:08.20 | Sato1 | ...and less live neurones |
| 14:08.25 | Zeeek | and likely three or four other langs as well |
| 14:08.55 | *** join/#asterisk user7621 (~ontae@chello213047229097.tirol.surfer.at) |
| 14:09.18 | _-Jon-_ | Hey I have a strange problem.. I'm using my ATA at home and dial out, I hear the ringing sound. But if I dial into the system with my cell phone, or any phone for that matter, and dial an extention that goes out the same line, I don't hear ringing. Any ideas? |
| 14:09.33 | *** join/#asterisk mogorman (~mogorman@207.111.174.1) |
| 14:09.41 | vaewyn | I'll take a wifi over those @#$@#$ 2.4ghz cordless headsets |
| 14:09.56 | vaewyn | The wifi ones are <50mw |
| 14:10.11 | ManxPower | Heritic! |
| 14:10.36 | vaewyn | Some of the cordless ones but 450+mw out because they have @#$#$ antennas |
| 14:11.22 | vaewyn | ManxPower: try a Hitachi Cable WIP-5000 and see if you come back :P |
| 14:11.34 | ManxPower | vaewyn: It has 2 lines? |
| 14:11.34 | Sato1 | or i would have lot of colissions in my wireless internet link |
| 14:11.40 | newl | ditto..too much already in the air here |
| 14:11.51 | ManxPower | vaewyn: It's under $100? |
| 14:11.53 | vaewyn | ManxPower: Nope... callwaiting... but not 2 lines |
| 14:11.59 | vaewyn | 300$ |
| 14:12.12 | vaewyn | but your 2.4 + the SIP brick weren't under 100$ either |
| 14:12.14 | ManxPower | HAHAHAHA! |
| 14:12.37 | Sato1 | ManxPower, is there a patch to compile those codecs from Eric in asterisk? :D |
| 14:12.43 | vaewyn | plus mine works in airports... hotel rooms... anywhere there is wifi |
| 14:12.48 | vaewyn | :} |
| 14:12.50 | ManxPower | Sato1: No. I said don't download it. |
| 14:12.53 | cursor | DECT isn't 2.4G in England |
| 14:13.06 | vaewyn | Heck... I made calls from the VON floor with it :P |
| 14:13.18 | ManxPower | You can't expect me to help you on something I told you not do download. |
| 14:13.19 | Sato1 | oops, i thought you was being sarcastic |
| 14:13.19 | Sato1 | erases |
| 14:13.30 | vaewyn | also makes a great wifi network spotter :P |
| 14:13.36 | ManxPower | Sato1: No. I just don't want to answer questions about it. |
| 14:13.59 | Sato1 | ok |
| 14:14.20 | ManxPower | I donnated the entire file to asteriskdocs.org. |
| 14:15.00 | *** part/#asterisk user7621 (~ontae@chello213047229097.tirol.surfer.at) |
| 14:15.09 | ManxPower | Sato1: Like the Devil, I am known by many names. |
| 14:15.19 | Sato1 | hehehe |
| 14:15.23 | newl | Natas! |
| 14:15.26 | ManxPower | In OKC I'm known as "Shut up!" |
| 14:15.36 | cursor | :-) |
| 14:15.39 | AgiNamu | ManxPower, you're JerJer? |
| 14:15.53 | AgiNamu | ;) |
| 14:16.08 | ManxPower | AgiNamu: No. I would have to own guns and support the war to be JerJer |
| 14:16.16 | jskcr|lappy | s |
| 14:16.25 | cursor | haha |
| 14:16.30 | _-Jon-_ | Can someone tell me why this produces no ringing sound: exten => 1,1,Dial(SIP/2423,20,Ttr) |
| 14:16.40 | cursor | earplugs? |
| 14:16.58 | _-Jon-_ | cursor, was that directed towards me? |
| 14:17.03 | cursor | :-) |
| 14:17.15 | _-Jon-_ | I know I forgot to take those out! :P |
| 14:17.18 | _-Jon-_ | *knew |
| 14:17.27 | newl | *pop* |
| 14:17.37 | *** join/#asterisk darth-timeus (darth@200.105.128.61) |
| 14:17.56 | cursor | May the source be with you, Darth |
| 14:18.31 | cursor | Jon: Try without the Tt |
| 14:18.43 | cursor | the "r" should fake a ringtone |
| 14:18.50 | RoyK | friday...... |
| 14:18.57 | RoyK | 16:18 |
| 14:18.58 | cursor | I don't know how the Tt would affect it - I don't use Tt |
| 14:18.58 | ManxPower | cursor: as far as I can tell "r" never actually works when you need it to. |
| 14:19.04 | cursor | I don't use "r" either :-) |
| 14:19.05 | RoyK | 18 past weekend start |
| 14:19.08 | RoyK | 18minutes that is |
| 14:19.10 | *** join/#asterisk makhtar (~ageller@mail3.bulletinnews.com) |
| 14:19.10 | RoyK | 19 |
| 14:19.13 | RoyK | :D |
| 14:19.19 | _-Jon-_ | I like to have that assuring ring sound :P |
| 14:19.19 | cursor | 3:19pm here |
| 14:19.20 | ManxPower | You will hear ringing if Asterisk things you should hear ringing. |
| 14:19.30 | RoyK | cursor: sounds like uk |
| 14:19.39 | cursor | UK, yes |
| 14:19.42 | ManxPower | "r" tells Asterisk to provide ringing sound even when it thinks it's the wrong thing to do. |
| 14:19.44 | cursor | Is there anywhere else? |
| 14:19.49 | fenlander | :-) |
| 14:19.55 | RoyK | madeira? |
| 14:20.02 | newl | Australia? B) |
| 14:20.08 | Jas_Williams | No UK |
| 14:20.08 | RoyK | wrong tz :P |
| 14:20.10 | cursor | You can get a ring sound, followed by an engaged tone, which is just weird |
| 14:20.15 | cursor | Don't use it unless you have to |
| 14:20.16 | darth-timeus | hi |
| 14:20.40 | Jas_Williams | cursor is corrct do not use r unless you have a broken phone/gateway |
| 14:20.44 | darth-timeus | i can't get the asterisk to have two way audio |
| 14:20.56 | darth-timeus | i'm open to suggestion |
| 14:21.06 | Jas_Williams | darth-timeus: Nat or codec issue ? |
| 14:21.15 | ManxPower | I thought UK fell into the ocean! http://www.theregister.co.uk/2005/04/29/google_bush_map/ |
| 14:21.28 | AgiNamu | Can someone explain why it appears that x,y are used as loop counters in res_agi? |
| 14:21.37 | newl | No, it fell into his lap, along with Australia. ;) |
| 14:21.37 | darth-timeus | i don't have nat, or firewall configured |
| 14:21.40 | ManxPower | Sorry, that was the REST of Europe that fell into the ocean. |
| 14:21.46 | cursor | http://www.bushorchimp.com/ |
| 14:21.47 | AgiNamu | instead of i,j (or G-d forbid, descriptive names) |
| 14:22.00 | darth-timeus | Jas_Williams: i'm new to asterisk, how can i check if is a codec problem? |
| 14:22.08 | Zeeek | AgiNamu converted COBOL ? |
| 14:22.36 | Zeeek | darth-timeus disallow=none, allow=ulaw |
| 14:22.46 | Nuxi | mod_cobol:: typeorg physical-sequential |
| 14:22.50 | ManxPower | Zeeek: it's disallow=all |
| 14:22.50 | Zeeek | darth-timeus disallow=all, allow=ulaw |
| 14:22.52 | _-Jon-_ | Damn, still no ringing sound :( |
| 14:23.00 | Zeeek | I know, it's a funny hour for my typing |
| 14:23.08 | cursor | brb... |
| 14:23.17 | Zeeek | clean out the ringer, maybe it's clogged? |
| 14:23.23 | Jas_Williams | _-Jon-_: Are you calling through a gateway ? |
| 14:23.48 | *** join/#asterisk santiago (~santiago@63.245.86.227) |
| 14:23.48 | _-Jon-_ | Jas_williams, I'm calling a toll-free number from LiveVoip first. Is that what you mean? |
| 14:23.52 | Zeeek | I always use 'r' and I always have ringing |
| 14:24.06 | Zeeek | but once I had a distorted double ringing |
| 14:24.06 | newl | You've got too much equipment on the line which is beyond the max REN value. 8) |
| 14:24.22 | _-Jon-_ | Zeeek, I get that once in a while too |
| 14:24.37 | darth-timeus | Zeeek: it is configured in that way |
| 14:24.41 | Zeeek | it's an ugly sound, like two rings on top of each other |
| 14:24.58 | Zeeek | darth-timeus you need to pastebin an example of a call |
| 14:25.04 | Jas_Williams | _-Jon-_: are you tlking about the ring back tone generated by you phone when you make an outbound call via LiveVoip ? |
| 14:25.06 | Zeeek | or at least describe one |
| 14:25.09 | AgiNamu | well, there's a nested for loop, using x,y |
| 14:25.15 | AgiNamu | so I'm going to go rename that |
| 14:25.37 | AgiNamu | esp. cause it's unrelated to x,y coordinates :P |
| 14:25.37 | Zeeek | why do we always start with i in c, anyway? |
| 14:25.41 | _-Jon-_ | Jas_williams, right. I hear nothing unil they pick up |
| 14:26.05 | AgiNamu | Zeeek, cause thats just how we do it. |
| 14:26.08 | newl | Zeeek: because programmers are self centered. :) |
| 14:26.13 | AgiNamu | the thing is, you shouldnt ever get to k |
| 14:26.25 | AgiNamu | if you need 3 loop counters, some would say, if you ever get to j |
| 14:26.35 | AgiNamu | then you should rename to something useful, like ixCommand or ixArgument |
| 14:26.50 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:26.50 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:26.53 | Zeeek | newl then should it be "for (me=0;me<100;me++) ? |
| 14:27.02 | Jas_Williams | _-Jon-_: That will be a Live Voip problem they need to provide progress tones to * an r on your outgoing dial should force you phone to hear ringing but it is a nasty hack |
| 14:27.06 | AgiNamu | "Excess Flood"? Isn't the definition of Flood to do with Excess? |
| 14:27.12 | Zeeek | ADD A,B ; add a to b |
| 14:27.15 | AgiNamu | "Insufficient Flood" |
| 14:27.47 | Jas_Williams | Just Flood would be better English |
| 14:28.00 | *** join/#asterisk darth-timeus (darth@200.105.128.61) |
| 14:28.12 | AgiNamu | but im guessing that I'll be murdered if I try to submit a patch that uses "ixCommand" |
| 14:28.17 | Jas_Williams | darth-timeus: Use pastebin :-) |
| 14:28.18 | _-Jon-_ | Jas_Williams, err wait, hold on. Let me explain something else.. I dial my toll free number and my Asterisk box picks up. I dial the extention of my ATA at home and I hear no ringing either |
| 14:28.20 | Jas_Williams | ~opastebin |
| 14:28.50 | Jas_Williams | _-Jon-_: Whic version of asterisk are you running ? |
| 14:28.50 | *** join/#asterisk alvis (Alvis@200.105.128.59) |
| 14:28.51 | darth-timeus | Jas_Williams: how i do that? |
| 14:29.08 | Zeeek | -Jon- We do that all the time. Who's your DID from? |
| 14:29.17 | AgiNamu | Is there any general prefix that's most Asterisk acceptable? like curcmd or curarg? |
| 14:29.27 | Zeeek | ~pastebin |
| 14:29.28 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 14:29.31 | _-Jon-_ | Zeeek, Livevoip |
| 14:29.45 | _-Jon-_ | Jas_Williams, not sure, one of the CVS versions |
| 14:29.50 | Zeeek | and haven't there been msgs on the mailing list about them recently? |
| 14:30.17 | darth-timeus | ~pastebin |
| 14:30.18 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 14:30.38 | Jas_Williams | _-Jon-_: Do a make update in the src directory and try again :) or do a show version in the cli |
| 14:30.50 | Zeeek | -Jon- type this into google: livevoip ring site:lists.digium.com |
| 14:31.05 | Zeeek | or is that your message? |
| 14:32.08 | *** part/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 14:32.16 | _-Jon-_ | Zeeek, wouldn't the problem be with my Asterisk box though? Cause it's not rining when I call into Asterisk through LiveVoip, and then dial any extension |
| 14:32.22 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 14:32.38 | _-Jon-_ | Jas_williams, doesn't give me a version # or date |
| 14:32.40 | ManxPower | _-Jon-_: That is a known problem. No fix that I know of. |
| 14:32.41 | _-Jon-_ | I'll try updating |
| 14:32.53 | ManxPower | ~google site:lists.digium.com livevoip ringing |
| 14:33.05 | Zeeek | http://lists.digium.com/pipermail/asterisk-users/2005-April/100653.html |
| 14:33.11 | _-Jon-_ | ManxPower, oh okay, so not a problem on my part then |
| 14:33.24 | ManxPower | _-Jon-_: there are Results 1 - 10 of about 76 from lists.digium.com for livevoip ringing. |
| 14:33.41 | Jas_Williams | _-Jon-_: How about when you do an asterisk -r should show version as the first few lines |
| 14:33.47 | Jas_Williams | root@asterisk:~# asterisk -r |
| 14:33.47 | Jas_Williams | Asterisk CVS-HEAD-05/12/05-09:05:09, Copyright (C) 1999 - 2005 Digium. |
| 14:33.47 | Jas_Williams | Written by Mark Spencer <markster@digium.com> |
| 14:34.01 | cursor | Mine doesn't say that :-) |
| 14:34.05 | ManxPower | It's one of several reasons I won't use LiveVoIP. |
| 14:34.15 | _-Jon-_ | Asterisk , Copyright (C) 1999 - 2005 Digium. |
| 14:34.23 | Zeeek | show version |
| 14:34.34 | _-Jon-_ | Asterisk built by root@xero on a i686 running Linux |
| 14:34.35 | _-Jon-_ | :P |
| 14:34.44 | ManxPower | Another reason is that they are a small company that cares about it's customers about as much as a large customer. |
| 14:34.55 | Jas_Williams | Ah its broke then :) |
| 14:34.59 | _-Jon-_ | Heheh |
| 14:35.05 | _-Jon-_ | Time to get newest CVS :P |
| 14:35.32 | eper-werk | ill have a asterisk pbx fully funning if voiptalk every send the card out |
| 14:35.48 | Jas_Williams | Do a make update in the src directory then make install to get latest cvs version and update version flags |
| 14:35.49 | darth-timeus | ok here is the pastebin |
| 14:35.51 | darth-timeus | http://pastebin.ca/11661 |
| 14:35.57 | cursor | I haven't taken the plunge back to CVS HEAD yet - still on CVS v1-0, but up to date |
| 14:36.03 | _-Jon-_ | How often do you guys rebuild Asterisk? |
| 14:36.12 | *** join/#asterisk flynux (durg9mz@pingou.in) |
| 14:36.25 | cursor | Whenever v1-0 updates come in on the asterisk-cvs mail list |
| 14:36.35 | cursor | haha |
| 14:36.43 | _-Jon-_ | Hehe |
| 14:37.18 | cursor | haha |
| 14:37.22 | cursor | Not smelling like tuna |
| 14:37.28 | Zeeek | darth-timeus I don't think you mention this was h323? |
| 14:37.48 | Jas_Williams | darth-timeus: Which version of asterisk are you using and which h323 stack ? |
| 14:37.51 | darth-timeus | Zeeek: i'm so sorry, my mistake |
| 14:37.59 | Zeeek | it's an important element |
| 14:38.11 | cursor | The H.323 standard was written by a Brontosaurus |
| 14:38.31 | Zeeek | I won't value-judge, but I've neverused it ever so I can't be any use here |
| 14:38.44 | Zeeek | like asking your priest for sex advice |
| 14:38.50 | darth-timeus | i'm using the last cvs, with openh323 v1_17_1 |
| 14:38.55 | Zeeek | these days, maybe just asking for sex |
| 14:39.14 | darth-timeus | jajaja |
| 14:39.20 | cursor | Perhaps if you're a young boy and ask a Catholic priest |
| 14:39.51 | Dishwasha | what do you mean these days, Catholic priest have been molesting young boys for centuries |
| 14:40.07 | Jas_Williams | darth-timeus: which channel driver chan_openh323 or chan_h323 ? |
| 14:40.17 | cursor | haha |
| 14:40.27 | darth-timeus | Jas_Williams: chan_h323 |
| 14:40.34 | Zeeek | the latest craze is female teachers asking 12 year old boys |
| 14:40.43 | Zeeek | shit where were they 30 years ago? |
| 14:40.49 | newl | lucky boys :) |
| 14:41.07 | cursor | haha |
| 14:41.17 | *** part/#asterisk alvis (Alvis@200.105.128.59) |
| 14:41.41 | Dishwasha | See what happens when you're not allowed to spank anybody? |
| 14:41.46 | cursor | oops - scaring the locals |
| 14:41.49 | *** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
| 14:42.11 | Dishwasha | The spanking used to get the teachers' sexual frustrations out, but now they have to seek an alternative way |
| 14:42.29 | clint_ | Has anyone here implemented or seen something like "Remote Access to Call Forwarding" feature offered by the phone co? |
| 14:42.54 | Jas_Williams | darth-timeus: Could you post a h323 debug to pastebin |
| 14:43.03 | darth-timeus | ok |
| 14:43.10 | Zeeek | clint_ you mean like being able to control what numbers are called when a call comes in? |
| 14:43.23 | cursor | like "follow me?" |
| 14:43.36 | Zeeek | no, "walk this way" |
| 14:43.42 | Jas_Williams | darth-timeus: is your h323 device on the same subnet as * ? |
| 14:43.43 | newl | talk this way |
| 14:43.47 | cursor | Ministry of Silly Walks |
| 14:43.49 | Zeeek | app_walk_this_way.so |
| 14:44.36 | clint_ | Zeeek: The phone company gives a number you dial, you enter your phone number, a passcode, and where you want your calls delivered... |
| 14:44.55 | cursor | You could do that with an AGI/app and use the Asterisk database |
| 14:45.05 | newl | I've done all sorts of call control extensions like that using RT. It's a piece of cake. |
| 14:45.13 | Zeeek | clint_ look up DISA and astdb |
| 14:45.15 | cursor | cake - mmmmm |
| 14:45.25 | Zeeek | we do it all the time |
| 14:45.57 | *** part/#asterisk santiago (~santiago@63.245.86.227) |
| 14:46.23 | darth-timeus | Jas_Williams: here is the pastebin http://pastebin.ca/11662 |
| 14:46.30 | Zeeek | clint_ you use a PutDB to save a number, and then check the variable each time a call is handled |
| 14:46.44 | darth-timeus | Jas_Williams: yes |
| 14:46.46 | clint_ | Yeah, that was my first inclination (dialplan hack using the db) but was wondering if anyone had a solution in place or any experience with what to avoid... |
| 14:47.06 | cursor | Just be sure to password it |
| 14:47.09 | _-Jon-_ | Isn't this strange.. I just downloaded newest CVS and it still shows no version |
| 14:47.16 | cursor | The VM pin will probably do |
| 14:47.28 | Zeeek | clint_ it's a common example in docs |
| 14:47.35 | clint_ | Yeah, that was my next issue - is there an easy way to have it look at their voicemail password? |
| 14:47.44 | cursor | yes |
| 14:47.56 | newl | hmm..that DISA is intended for something similar to a remote call control of facilities. |
| 14:48.02 | clint_ | Ok, I'll bite... |
| 14:48.11 | cursor | :-) |
| 14:48.39 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) |
| 14:48.46 | MattH | Does anyone have any advice on getting voipjet to work? |
| 14:48.51 | clint_ | I still have voicemail using the flat file for configs, so the password is in there.. is there an easy way to access that from the dialplan? |
| 14:48.54 | MattH | I'm having an issue making any outgoing calls |
| 14:49.35 | _-Jon-_ | Hey I figured out a way around the ringing problem.. Set your hold music as a ring tone and use m instead of r :) |
| 14:49.37 | Zeeek | MattH what's wrong, it works AFAIK? |
| 14:49.37 | newl | clint_: that IMO should remain as seperate subsystems. |
| 14:50.06 | Nuxi | clint_, agi is the answer to all questions for which 42 is not appropriate. |
| 14:50.07 | newl | clint_: As a subscriber, you may wish someone to have access to voicemail but not to your CF* facilities. |
| 14:50.09 | clint_ | newl: and it will, cause I don't think there is an Easy Way(tm) :) |
| 14:50.30 | *** join/#asterisk HA (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
| 14:52.05 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
| 14:52.12 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 14:52.56 | clint_ | newl: Yeah, and I think it's more graceful than the 50 or so line dialplan hack I have cobbled together so far this morning.... |
| 14:53.30 | newl | clint_: it'll get larger as you start adding more facilities, trust me. :) |
| 14:53.37 | Zeeek | lookup "follow me" on wiki etc, there are several of these around |
| 14:53.39 | darth-timeus | Jas_Williams: how do i switch from chan_h323 to chan_openh323 |
| 14:53.48 | cursor | clint_: VMAuthenticate() |
| 14:54.08 | *** join/#asterisk _omer (dfsdf@202.147.174.176) |
| 14:54.10 | newl | I'm up to 474 lines in my facilities conf file alone. |
| 14:54.15 | clint_ | newl: My thoughts exactly. I look at what I have here and think "ok, now I need to add the ability to modify some other parameters from here...." Ugh. |
| 14:54.45 | clint_ | newl: Not to mention the stdext macro is getting out of hand. |
| 14:55.20 | _omer | Installed Linux - - - Installed Asterisk - - - now do I have to compile G729 (Which is available for testing) or what? |
| 14:55.22 | Zeeek | Personally, I use groups of short commands like changing the number of rings, whaich lines are answered etc |
| 14:55.24 | clint_ | What we need now is a function DoCLASS() :) |
| 14:55.34 | zoa | hey kram |
| 14:55.37 | bkw_ | _omer, No you can't |
| 14:55.44 | _omer | I cant what? |
| 14:55.44 | newl | one of the first things that happen in both incoming and outgoing contexts is that things are directed through the facility extensions. |
| 14:55.46 | bkw_ | _omer, what country you in? |
| 14:55.50 | kram | greets zoa babe |
| 14:56.07 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 14:56.21 | clint_ | On another note, anyone here with Q.931 expertise? I've got a helluva dispute going on with my carriers over some seemingly broken ISDN D channel behavior... |
| 14:56.23 | blitzrage | all hail kram! :) |
| 14:56.25 | _omer | Asterisk is in USA ....and I'm in Pakistan |
| 14:56.25 | cursor | Pakistan |
| 14:56.44 | MattH | Zeeek: sorry was distracted.. I get an: Called voipjet/number Hangup IAX2/voipjet/2 |
| 14:56.55 | bkw_ | _omer, you can't compile g729 you must purchase it to use it |
| 14:57.07 | Zeeek | MattH I just dialed a number and it works fine |
| 14:57.15 | _omer | why???? is it because of country or what? |
| 14:57.24 | MattH | my config is at: http://pastebin.com/283694 |
| 14:57.26 | bkw_ | you must pay.. its covered by patents |
| 14:57.27 | MattH | in iax2.conf |
| 14:57.29 | blitzrage | _omer: its a proprietary codec |
| 14:57.37 | cursor | It's closed source |
| 14:57.43 | blitzrage | ^^^ |
| 14:57.50 | _omer | yes I know...but for testing purpose it is available in Asterisk...isn't it? |
| 14:57.56 | blitzrage | no! |
| 14:57.56 | bkw_ | nope |
| 14:58.00 | blitzrage | its closed source |
| 14:58.12 | blitzrage | Asterisk will do passthrough if both ends are using that codec afaik |
| 14:58.30 | ManxPower | _omer: The G729 that you are referring to is illegal in most of the world. |
| 14:58.33 | MikeJ[Laptop] | _omer. the licences are inexpensive, just buy a couple... |
| 14:58.35 | Zeeek | MattH no disallow=all ? |
| 14:58.47 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 14:58.51 | blitzrage | isn't it only like $5 a license? |
| 14:58.57 | *** join/#asterisk jmacz (~jmacz@63.245.86.192) |
| 14:58.57 | AgiNamu | $10 from Digium |
| 14:58.58 | Zeeek | I also have dropcount=1 although I don't remember why now |
| 14:58.59 | AgiNamu | not bad at all |
| 14:59.01 | cursor | more like $10 |
| 14:59.07 | blitzrage | whatever.. still cheap :) |
| 14:59.10 | AgiNamu | well, not bad, relatively to paying $50K |
| 14:59.11 | bkw_ | per channel |
| 14:59.19 | blitzrage | key point :) |
| 14:59.19 | ManxPower | ~astg723.1 |
| 14:59.22 | ManxPower | oops |
| 14:59.22 | _omer | but I remember.....last time someone compiled G729 in my server and told me that it is for testing purpose...and when I need to use it commercially then I have to buy it.... |
| 14:59.26 | blitzrage | hell... just use something else |
| 14:59.28 | MattH | sorry also have |
| 14:59.28 | MattH | bindport = 4569 ; Port to bind to (IAX is 4569) |
| 14:59.28 | MattH | bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) |
| 14:59.28 | MattH | delayreject=yes |
| 14:59.28 | MattH | disallow=all |
| 14:59.28 | MattH | allow=ulaw |
| 14:59.29 | ManxPower | Here is the licensing priceing info for G723.1 direct from the patent holder's web site: http://www.dspg.com/technology/LicensePricing.html |
| 14:59.30 | MattH | allow=alaw |
| 14:59.33 | bkw_ | MattH, DO NOT DO THAT AGAIN |
| 14:59.37 | ManxPower | _omer: The lied. |
| 14:59.40 | AgiNamu | _omer, if you really want to, then you can go get the Intel sample code + ReadyTechnology open G729 |
| 14:59.41 | bkw_ | pastebin.ca is for that |
| 14:59.47 | AgiNamu | but it's "not legal" |
| 14:59.49 | MattH | ouch sorry botty |
| 14:59.50 | MikeJ[Laptop] | ~pastebin |
| 14:59.51 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 14:59.51 | AgiNamu | so no one here will help |
| 14:59.58 | blitzrage | bkw_: oh come on... its like... 7 lines |
| 15:00.05 | _omer | <ManxPower> : how come he lied....If I used that codec too.. |
| 15:00.06 | bkw_ | anything over 3 is too much |
| 15:00.08 | Zeeek | 6 lines |
| 15:00.13 | newl | If all 295 users pasted 7 lines... ;) |
| 15:00.16 | kram | 5.5 lines |
| 15:00.18 | MattH | LOL ok ok I repent |
| 15:00.19 | Zeeek | wait let's do that |
| 15:00.21 | ManxPower | _omer: perhaps he was confused. |
| 15:00.23 | blitzrage | bkw_: then I don't want to see you say more than 3 lines in a row :) |
| 15:00.24 | Zeeek | ready? |
| 15:00.30 | blitzrage | 1 |
| 15:00.31 | blitzrage | 2 |
| 15:00.31 | blitzrage | 3 |
| 15:00.32 | bkw_ | I usuallydon't |
| 15:00.33 | blitzrage | too much! |
| 15:00.43 | MattH | Zeeek: I'm not even seeing it registering... iax2 show registry? |
| 15:00.48 | bkw_ | yes it stoo much when people are talking and the lines get interweaved with other lines |
| 15:00.48 | AgiNamu | If all 295 users weren't away...... |
| 15:00.50 | ManxPower | People seem to think ANYTHING is legal if it's "just for testing" or "just for educational use" or "just for non-commercial use". That is simply not true. |
| 15:01.01 | blitzrage | bkw_: thats the whole point of IRC :) |
| 15:01.05 | MattH | wait I take that back.. it is registering.. hrmm |
| 15:01.08 | Zeeek | MattH you have a qualify= in the voipjet entry? |
| 15:01.12 | bkw_ | ManxPower, it might be ok for edu use... but not for comercial. |
| 15:01.17 | newl | ManxPower: It may be true if the country you live in permits it. |
| 15:01.29 | blitzrage | not permits, but doesn't enfore :) |
| 15:01.36 | bkw_ | ya |
| 15:01.37 | ManxPower | bkw_: I've yet to see ANYONE cite a legal basis for any of those uses. |
| 15:01.38 | Zeeek | 729 is free in ocuntries that don't have capital punishment |
| 15:01.50 | bkw_ | VoiceAge gives out a 1 channel g729 for testing |
| 15:02.01 | bkw_ | win32 only |
| 15:02.05 | bkw_ | but still its out there |
| 15:02.08 | ManxPower | newl: I *suspect* that the G729 stuff is covered under ITU Treaty, not patent laws. |
| 15:02.08 | Nuxi | what about uncial punishment? |
| 15:02.09 | MattH | Zeeek: ack... yes... now it's working.. hrmm wierd |
| 15:02.21 | bkw_ | nazi ITU |
| 15:02.42 | Zeeek | MattH follow the mailing list, there have been a few messages about voipjet (and others) |
| 15:02.49 | ManxPower | If it wasn't then there would be all sorts of companies selling unlicensed G729 phones in most of the world. |
| 15:02.56 | MattH | Zeeek: thanks |
| 15:03.01 | Zeeek | np |
| 15:03.22 | Zeeek | for those of you in Europe there is now voipbuster |
| 15:03.34 | *** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 15:03.51 | Zeeek | voipbuster/myname 213.61.187.154 (S) 255.255.255.255 4569 OK (20 ms) |
| 15:04.11 | Nugget | kickass. apple's support line is in e164, so when I call it goes straight there via sip. |
| 15:04.16 | _-Jon-_ | Anyone know of a simple way to do a callback? I tried setting up an extention that hangs up and then dials my cell phone but once it hangs up it just stops |
| 15:04.36 | Zeeek | Jon use a call file |
| 15:04.50 | _-Jon-_ | Call file? |
| 15:05.16 | Zeeek | <PROTECTED> |
| 15:05.22 | Zeeek | they call you |
| 15:05.27 | _-Jon-_ | Ohhh |
| 15:05.27 | ManxPower | _-Jon-_: Instead of saying "Call file?" Try asking the wiki "Call file?" |
| 15:05.47 | Zeeek | <PROTECTED> |
| 15:05.53 | _-Jon-_ | ManxPower, true, the wiki is smarter :) |
| 15:05.54 | _-Jon-_ | jk |
| 15:06.05 | ManxPower | Zeeek: Did he at least buy you dinner and drinks before you start holding his hand? |
| 15:06.10 | Zeeek | haha |
| 15:06.13 | _-Jon-_ | haha |
| 15:06.18 | Zeeek | I have a need for increased karma |
| 15:06.33 | ManxPower | I, of course, require dinner, drinks, AND cash before doing any handholding. |
| 15:06.37 | Zeeek | speaking of which, when/if you come to Paris, I'll buy you lunch |
| 15:06.47 | Zeeek | but I don't do it on the first date |
| 15:07.00 | _-Jon-_ | ManxPower, name your price |
| 15:07.06 | cursor | ManxPower: Don't forget the film |
| 15:07.14 | ManxPower | Zeeek: If nobody else asks to meet me regarding potential jobs I'll be spending a day or so in Paris. |
| 15:07.45 | Zeeek | in that case, you'd be better to go Amst. Paris MAdrid I think, no? |
| 15:07.48 | cursor | ugh |
| 15:07.53 | cursor | Paris is full of French people |
| 15:07.58 | cursor | :-) |
| 15:07.58 | Zeeek | Amsterdam-Paris is 5 hours |
| 15:07.59 | ManxPower | Zeeek: Yes. |
| 15:08.14 | Zeeek | let me know: manx-zeeek@sneakemail.com |
| 15:08.20 | _-Jon-_ | Heh they have REAL french people there |
| 15:08.23 | _-Jon-_ | Unlike Canada |
| 15:08.50 | _-Jon-_ | No offence to any Quebecers :) |
| 15:08.53 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
| 15:08.55 | ManxPower | I got new glasses this week so I'm not the Hip Geek Manx Power |
| 15:09.14 | _-Jon-_ | Are there even any French people in BC? |
| 15:09.21 | Zeeek | no |
| 15:09.29 | Zeeek | that's why it was stupid |
| 15:09.32 | ManxPower | ..er. not == NOW |
| 15:09.34 | cursor | "She's French-Canadian. Sometimes she's Canadian and can be quite pleasent. Today, she's obviously French" |
| 15:09.39 | Nuxi | I'll take an order of French toash with a side of French fries with som French mustard. |
| 15:09.48 | cursor | I forget which film that's from |
| 15:09.53 | ManxPower | Zeeek: It's funny to see French people expect people in New Orleans to speak french. |
| 15:09.56 | _-Jon-_ | Hah |
| 15:10.07 | cursor | A film with a mountain in it |
| 15:10.18 | ManxPower | I think there are like 10 people in New Orleans that speak French. LOL! |
| 15:10.20 | Zeeek | ManxPower there are some people who do |
| 15:10.32 | Zeeek | Zydeco |
| 15:10.37 | cursor | Je parle un petit Français mais pas très bien. |
| 15:10.44 | cursor | And that's about it |
| 15:10.49 | ManxPower | Zeeek: I lived in New Orleans for 10 years and never heard a local person speak french. |
| 15:10.56 | Zeeek | *good enough to get someone in bed, that |
| 15:11.23 | Zeeek | I've never been to N.O. - always wanted to go |
| 15:11.24 | CyberKnet | ManxPower: why does you saying that not surprise me? |
| 15:11.28 | CyberKnet | ;) |
| 15:11.38 | cursor | Just say NO |
| 15:11.51 | ManxPower | Zeeek: New Orleans has under 1 million people, and 300 murders per year. |
| 15:11.54 | CyberKnet | cursor: amen to that. |
| 15:11.55 | *** join/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 15:12.03 | darwin35 | BV is back |
| 15:12.09 | ManxPower | I think the city itself is about 300,000 and has almost 300 murders per year. |
| 15:12.26 | Zeeek | one in 1000 murdered each year? |
| 15:12.33 | cursor | Live just outside the city then - 0 murders per year |
| 15:12.37 | ManxPower | Zeeek: Yes. |
| 15:12.37 | Zeeek | must all be geeks |
| 15:13.01 | Zeeek | what are the murder motives? crack? |
| 15:13.05 | *** join/#asterisk cmk (~cmk_@p54A3CFAE.dip.t-dialin.net) |
| 15:13.14 | cursor | Put an electric fence around the city and let them all kill one another |
| 15:13.19 | CyberKnet | Zeeek: more like people euthanizing =) |
| 15:14.02 | Zeeek | <PROTECTED> |
| 15:14.08 | ManxPower | Looks like the crime rate was down in 2004: http://www.cityofno.com/portal.aspx?portal=50&tabid=12 |
| 15:14.09 | Zeeek | Chicago |
| 15:14.11 | cursor | haha |
| 15:14.19 | cursor | probably less people to kill than last year |
| 15:14.22 | newl | hah those call files are nifty |
| 15:14.47 | ManxPower | Zeeek: 2/3 of the citizens of New Orleans over 18 are illiterate. That and poverty. |
| 15:15.00 | Zeeek | they should be too stupid to kill each other! |
| 15:15.12 | newl | That's the problem, they don't realize it. 8) |
| 15:15.19 | _-Jon-_ | hmmm so how can i activate this callfile from an extension? there is no mention of this anywhere |
| 15:15.20 | Zeeek | course, you don't need an IRC channel to compile a gun, do you? |
| 15:15.21 | ManxPower | 4468 violent crimes in New Orleans in 2004 |
| 15:15.45 | cursor | Jon: Why do you want to activate it from an extension? |
| 15:15.55 | cursor | if you want to call a phone - just call it |
| 15:15.59 | ManxPower | For that link is only the city of new orleans i.e. 300,000 people. |
| 15:16.06 | _-Jon-_ | cursor, heres my plan (it's to get free long distance :P) |
| 15:16.10 | Zeeek | -Jon- the short answer is system() - the long one, do a little reading! |
| 15:16.26 | _-Jon-_ | cursor, I call toll free number, press a button and it hangs up and calls me back using BroadVoice. Then I dial out |
| 15:16.26 | cursor | Like a callback? |
| 15:16.33 | cursor | ok |
| 15:16.44 | Zeeek | DISA will do that too without hanging up |
| 15:16.51 | Zeeek | Starter tutorial: |
| 15:16.51 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 15:16.51 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 15:16.51 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 15:16.51 | Zeeek | THE reference of the moment: |
| 15:16.52 | Zeeek | http://www.asteriskdocs.org |
| 15:16.55 | *** join/#asterisk cmk (~cmk_@p54A3CFAE.dip.t-dialin.net) |
| 15:17.02 | _-Jon-_ | Zeeek, you are quick :) |
| 15:17.10 | Zeeek | flood! flood! |
| 15:17.12 | Silik0n | last -msg |
| 15:17.17 | Nuxi | Zeek, next time use pastebin |
| 15:17.20 | Zeeek | no |
| 15:17.28 | Nuxi | ;) |
| 15:17.31 | *** join/#asterisk rooster27 (~rob@ool-18bdeca2.dyn.optonline.net) |
| 15:17.33 | Zeeek | I could put it all on one line but then it isn't readable |
| 15:18.14 | CyberKnet | Zeeek: yep. pastebin is good for multi-line content, plus as a bonus it stays around forever, so it will be indexed by google. |
| 15:18.17 | Zeeek | -Jon- the questions you are asking, we've all asked ourselves. There are lots of examples out there |
| 15:18.29 | Zeeek | pastebin is poo |
| 15:18.33 | cjk | is there any way to save the registration status (astdb) into mysql? |
| 15:18.36 | Zeeek | except for debug output and programs |
| 15:18.37 | CyberKnet | pastebin.ca rocks |
| 15:18.47 | cursor | Time for me to go |
| 15:18.51 | CyberKnet | later cursor |
| 15:18.53 | Zeeek | it should expire by default, too |
| 15:18.54 | cursor | I might be back later |
| 15:18.59 | cursor | If you're unlucky |
| 15:19.04 | Zeeek | thanks for the warning cursor :) |
| 15:19.13 | cursor | :-) |
| 15:19.13 | CyberKnet | Zeeek: pastebin.ca sets about to create a knowledgebase. expiring by default is counter-intuitive. |
| 15:19.15 | _-Jon-_ | Zeeek, hmm the .call file seems to be what I want but like I said, I googled it, checked the wiki and it seems like the .call file can only be used to call at a specific time |
| 15:19.17 | cursor | This message probably won't change while you're staring at it. |
| 15:19.23 | CyberKnet | cursor: =) |
| 15:19.26 | cursor | This message is slightly different than the one that was here a minute ago. |
| 15:20.01 | Zeeek | -Jon- you need to think a little then |
| 15:20.11 | Zeeek | your script will generate a .call file |
| 15:20.35 | Zeeek | type this: show application system |
| 15:20.55 | Zeeek | so you'll write a script named "callback" |
| 15:21.06 | _-Jon-_ | ohh okay |
| 15:21.18 | Zeeek | callback, in any language you like, will create a allback.call file |
| 15:21.28 | Zeeek | and move it to the spool/outgoing directory |
| 15:22.09 | Zeeek | exten => s,1,system(callback ${EXTEN} param1 hahaha) |
| 15:22.31 | Zeeek | except it won't be EXTEN for 's' |
| 15:22.33 | _-Jon-_ | ohh okay, you're smart zeeek :P |
| 15:22.43 | Zeeek | no it's all wrong, but that's the idea |
| 15:22.46 | ManxPower | I'll be in #asterisk-stable (where all the cool kids hang out) if any has questions about 1.0.x |
| 15:22.57 | Zeeek | I have a question about 1.0.6 |
| 15:23.11 | Zeeek | will it always be stable? |
| 15:23.44 | Juggie | argh... i dont understand why these sip clients ignore sdp |
| 15:24.05 | darwin35 | upgrade from 1.0.6 to 1.0.7 |
| 15:24.08 | *** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 15:24.21 | darwin35 | 1.0.6 was buggy |
| 15:24.29 | Zeeek | but it works! |
| 15:24.37 | Zeeek | I'm too scared to change |
| 15:27.20 | *** join/#asterisk wasabi_ (~wasabi@207.55.180.100) |
| 15:27.27 | wasabi_ | Does Asterisk allow you to monitor calls? |
| 15:27.49 | AgiNamu | yes. |
| 15:27.49 | zoa | yes |
| 15:28.07 | *** join/#asterisk Fanguin (~Fanguin@p548F1EE9.dip0.t-ipconnect.de) |
| 15:28.12 | wasabi_ | I'm trying to get my head around all this and I'm having a hard time. We have some phones with a MOnitor button |
| 15:28.15 | AgiNamu | think of asterisk as a call platform. call comes in, then you have your way with it. |
| 15:28.22 | wasabi_ | Do we just somehow program these buttons to do something? |
| 15:28.26 | wasabi_ | or do they just work? |
| 15:28.29 | wasabi_ | I don't understand the interaction. |
| 15:28.32 | HA | where is the best place to get plantronics non-amplified headsets for cheap? |
| 15:28.38 | AgiNamu | Asterisk doesn't necesarily understand anything about your phones. |
| 15:28.49 | wasabi_ | So how do we set up the phones? |
| 15:28.59 | darwin35 | you unbox it |
| 15:29.00 | AgiNamu | I'm not sure what the monitor button on the phone does. |
| 15:29.00 | HA | what kind of phone is it wasabi? |
| 15:29.11 | darwin35 | plug it in to the power and the net |
| 15:29.22 | darwin35 | and cross your fingers |
| 15:29.24 | wasabi_ | Well, on our current (sucky system) it asks what extension the user wants to listen into |
| 15:29.29 | wasabi_ | without interrupting it |
| 15:29.32 | HA | what kind of phone is it wasabi? |
| 15:29.40 | wasabi_ | It's not voip. I'm just saying. |
| 15:30.06 | HA | do you intend to use a different phone with asterisk then? |
| 15:30.34 | wasabi_ | Yes. And I don't know what phone yet. |
| 15:30.39 | AgiNamu | Well, you can do that with Asterisk, but you're going to need some kind of way for all this to work |
| 15:30.40 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 15:30.59 | wasabi_ | I'm simply asking questions to get an understanding. |
| 15:31.04 | AgiNamu | If you think of Asterisk as a platform, like Linux or Apache |
| 15:31.07 | wasabi_ | We'll be buying a bunch of phones. They'll have buttons on them. |
| 15:31.15 | AgiNamu | that might help |
| 15:31.17 | wasabi_ | AgiNamu, So I have to write code to make everything I want work right? |
| 15:31.24 | HA | you can do the same with asterisk but you will need to configure asterisk with an extension that asks what extension you want to monitor. |
| 15:31.26 | AgiNamu | not necesarily, it does al ot of stuff out of the box |
| 15:31.29 | *** join/#asterisk devel (~devel@wiggum.digitalcoven.com) |
| 15:31.30 | Zeeek | wasabi_ if the buttons are programmable, you just write extensions to make asterisk do what you want |
| 15:31.38 | wasabi_ | "write extensions?" |
| 15:31.39 | AgiNamu | But if you want things to get monitored, then you tell Asterisk how you want that to work |
| 15:31.41 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 15:31.46 | AgiNamu | wasabi_, do this |
| 15:31.48 | AgiNamu | download asterisk |
| 15:31.52 | AgiNamu | then read all the sample config files. |
| 15:31.53 | wasabi_ | I have. |
| 15:31.55 | AgiNamu | that's a great start |
| 15:32.00 | wasabi_ | n't done that. |
| 15:32.05 | wasabi_ | Oh great. |
| 15:32.10 | wasabi_ | Is there any kinda GUI config util? |
| 15:32.14 | AgiNamu | that'll explain a lot of the features |
| 15:32.25 | HA | take a look at extensions.conf and get familiar with it. then you'll understand extensions. |
| 15:32.27 | AgiNamu | there are many GUIs |
| 15:32.27 | rooster27 | hey all, i've got an outbound call problem. when i try to call an external number, it tells me "we're sorry, our circuits are busy now. will you please try your call again later?" |
| 15:32.29 | AgiNamu | not sure if any don't suck. |
| 15:32.46 | Zeeek | rooster27 calling what? |
| 15:32.48 | AgiNamu | Asterisk does a shitload out of the box |
| 15:32.50 | rooster27 | my co-worker (who is in the same physical location as the phone server) can make outbound calls |
| 15:32.55 | AgiNamu | And it can do even more if you don't mind writing in C |
| 15:33.00 | rooster27 | but i'm at a remote location and i can't |
| 15:33.08 | wasabi_ | Well, see, I'm looking for a replacement to our current call center solution. |
| 15:33.19 | wasabi_ | I don't mind setting it up initially being a little techy |
| 15:33.26 | rooster27 | zeeek, when i dial 9 to try to hit an external number |
| 15:33.34 | rooster27 | and then i dial the number |
| 15:33.35 | wasabi_ | But if I have to tell the call center manager to get in there and edit text files thru ssh to add new extensions, well it ain't gonna work. ;0 |
| 15:33.35 | rooster27 | it says that |
| 15:33.36 | *** join/#asterisk ronn (ronn@host217-46-199-164.in-addr.btopenworld.com) |
| 15:34.04 | Zeeek | rooster27 could be a million things at this point |
| 15:34.15 | Zeeek | SIP? NAT? codec problems? |
| 15:34.23 | rooster27 | doh |
| 15:34.27 | rooster27 | well |
| 15:34.39 | Zeeek | what phone? what router setup? |
| 15:34.40 | wasabi_ | How are the extra buttons delt with at the protocol level? Does SIP have some sort of signal for "some button was pressed"? |
| 15:34.42 | rooster27 | my phone registers and i can call my co-worker on his extension - does that eliminate SIP ? |
| 15:34.54 | Zeeek | wasabi_ no which is why you need extensions |
| 15:34.57 | rooster27 | it's a Cisco 7960 behind a NAT'ed cable modem connection |
| 15:35.11 | Zeeek | rooster27 and are there ports forwxarded? |
| 15:35.37 | Zeeek | is the number you are trying to call on PSTN or SIP provider? |
| 15:35.47 | rooster27 | PSTN |
| 15:35.55 | Zeeek | wasabi_ see the button can be programmed to call a number |
| 15:36.21 | Zeeek | rooster27 you are sure there are available lines? |
| 15:36.26 | HA | where is the best place to get plantronics non-amplified headsets for cheap? |
| 15:36.37 | Zeeek | Midnight Headsets? |
| 15:36.49 | wasabi_ | Oh. Is progamming the button a per-phone thing? |
| 15:36.53 | rooster27 | zeeek, we have 5 lines plugged into the system and it's just me and my co-worker using it. i made sure he was off the phone when i tried |
| 15:36.55 | wasabi_ | I think I get it then. |
| 15:36.59 | Zeeek | wasabi_ phone dependent |
| 15:37.02 | wasabi_ | Asterisk holds an extension, running a "monitor script" |
| 15:37.09 | wasabi_ | You program the phone button to call it. |
| 15:37.12 | wasabi_ | ? |
| 15:37.19 | Zeeek | yes that's it |
| 15:37.21 | AgiNamu | you want to spy on another channel |
| 15:37.23 | HA | yep, thats about right. |
| 15:37.36 | Zeeek | funny, SIP should have had a button system in it, like MIDI does |
| 15:37.46 | HA | i use chan_spy to monitor agent channels. its very easy to do. |
| 15:37.46 | eper-werk | whats the -r switch do with sed on linux? |
| 15:38.00 | wasabi_ | how do I get to "my friendly asterisk CLI prompt?" =) |
| 15:38.08 | AgiNamu | The Monitor cmd will actually save the current call to a file. |
| 15:38.22 | Zeeek | ZapBarge is what you want |
| 15:38.32 | Zeeek | if it's ZAP |
| 15:38.48 | *** join/#asterisk cmaj (~chris@65-37-6-42.nrp2.roc.ny.frontiernet.net) |
| 15:38.55 | AgiNamu | ChanSpy won't allow you to enter an extention to listen to |
| 15:39.07 | wasabi_ | What is this "zap" thing? |
| 15:39.12 | HA | or chan_spy, if your using head, will do the same thing. |
| 15:39.14 | AgiNamu | zap refers to zapata telephony |
| 15:39.20 | *** join/#asterisk boch (~as24@200.59.172.98) |
| 15:39.24 | Zeeek | wasabi_ time to do some reading |
| 15:39.24 | AgiNamu | interfaces to T1 lines or analog telephone lines. |
| 15:39.32 | AgiNamu | You will most likely have to keep trck of which extension is using which channel to use chanspy |
| 15:39.34 | *** join/#asterisk jeffreyeni (meltzer@exobit.exobit.org) |
| 15:39.37 | HA | yeah, you have to do some extra work to get chanspy to ask for an extension. |
| 15:39.54 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 15:39.54 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 15:40.08 | Zeeek | wasabi_ time to do some reading ^^^^^^^^^^ |
| 15:40.19 | *** join/#asterisk jets (~brian@guardian.pmt.org) |
| 15:40.34 | HA | i use an agi to get the extension to listen to then i call chanspy using ChanSpy(|g) to get that particular SIP channel. |
| 15:40.45 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 15:42.12 | *** join/#asterisk joaovianna (joao@node-40247a6a.ewr.onnet.us.uu.net) |
| 15:43.41 | *** join/#asterisk alexns (~alex@acs-24-154-114-15.zoominternet.net) |
| 15:43.55 | robeni | Zeeek, if i can call my co-worker from my remote location, does that eliminate the possibility of a NAT or SIP issue? |
| 15:44.21 | alexns | Using TDM 400, asterisk 1.07 rxgain/txgain on fxo channels doesnt seem to work |
| 15:44.28 | alexns | anyone else seen this? |
| 15:45.42 | robeni | (my co-worker is also using the same phone system - not PSTN) |
| 15:46.00 | Zeeek | robeni it would seem to - is the coworker using SIP too? |
| 15:46.21 | robeni | yes, he's using SIP, registering to the same asterisk server |
| 15:46.46 | robeni | but somehow he can make outbound calls to PSTN numbers and i can't |
| 15:46.52 | Zeeek | are you configured for NAT (nat=yes, canreinvite=no) ? |
| 15:47.08 | Zeeek | and are ports forwarded? |
| 15:47.18 | Zeeek | and is asterisk also behind NAT? |
| 15:47.56 | Aze`DEV | Anyone know i cmd System() can return value ? |
| 15:47.59 | Zeeek | and what does the CLI say during the call? |
| 15:48.29 | Zeeek | Aze show application system will tell you |
| 15:48.39 | Zeeek | what it can return |
| 15:49.03 | AgiNamu | Aze`DEV, no, i think nothing can return vlaues to thedialplan |
| 15:49.28 | robeni | Zeeek, nevermind - i figured out the problem |
| 15:49.30 | Zeeek | if you write your own app you can set variables |
| 15:49.30 | robeni | i'm an idiot |
| 15:49.35 | robeni | thanks for your help |
| 15:49.39 | Zeeek | don't be too hard on yourself |
| 15:49.52 | *** join/#asterisk mountie (~mountie@24.42.99.232) |
| 15:50.19 | Aze`DEV | AgiNamu, i need to do "wget" and read text on grabbed page.. how ? agi->perl ? can u help me ? |
| 15:50.35 | Zeeek | robeni so tell us! |
| 15:51.07 | AgiNamu | AGI can set channel variables. |
| 15:51.27 | jaiger | hey all |
| 15:51.55 | robeni | the pbx recently moved from the same area code that i'm in to a different one |
| 15:51.56 | robeni | so |
| 15:52.02 | Zeeek | heh |
| 15:52.02 | robeni | now i need to include the area code when i dial an external # |
| 15:52.04 | robeni | heh |
| 15:52.04 | Aze`DEV | AgiNamu, are u crazy ? http://www.voip-info.org/wiki-set+variable |
| 15:52.05 | robeni | =) |
| 15:52.13 | robeni | <- dummy |
| 15:52.21 | Zeeek | shit happens |
| 15:52.45 | AgiNamu | Aze`DEV, wtf? |
| 15:56.38 | *** part/#asterisk jeffreyeni (meltzer@exobit.exobit.org) |
| 15:57.05 | *** part/#asterisk robeni (~rob@ool-18bdeca2.dyn.optonline.net) |
| 15:58.08 | Juggie | sigh |
| 15:58.10 | Juggie | i hate sip so much |
| 15:58.25 | Juggie | why do we have a damn sdp if we are going to ignore it |
| 15:58.54 | boch | why do you hate sip? |
| 15:58.59 | boch | sip is nice |
| 15:59.14 | Juggie | try firewalling around sip |
| 15:59.22 | CyberKnet | errr |
| 15:59.33 | CyberKnet | as in with a SIP ata behind a firewall? |
| 15:59.52 | Juggie | no, as in a asterisk server behind a firewall. |
| 16:00.03 | CyberKnet | ah. |
| 16:00.07 | CyberKnet | Haven't done that yet. |
| 16:00.12 | Zeeek | I have |
| 16:00.15 | Juggie | sdp tells the client, use 11527 or something to send me rdp |
| 16:00.25 | Juggie | and the client goes, screw you, i'm doing it my way |
| 16:00.32 | *** join/#asterisk |Vulture| (~V@161.233.204.68.cfl.res.rr.com) |
| 16:00.33 | CyberKnet | wow. that client is dumb. |
| 16:00.44 | Juggie | every client does it though... |
| 16:00.46 | Juggie | xten, sjphone |
| 16:00.51 | CyberKnet | what about firefly? |
| 16:01.03 | Juggie | they all try and be smart and use one udp connection |
| 16:01.33 | Juggie | eg, they receive packets from source port 3381: to 8000 rtp |
| 16:01.39 | Juggie | that would be in the case of say, xten |
| 16:01.46 | Juggie | so they go well, i'll just send back to the source port |
| 16:01.54 | newl | My Asterisk runs behind nat firewall just fine. |
| 16:01.54 | Juggie | so they send the rtp back to 3381 |
| 16:01.58 | CyberKnet | firefly has a box in options that says RTP Port that defaults to 5000 |
| 16:02.30 | Juggie | right, but the clients are returning the rtp to the source port that they received from |
| 16:02.38 | Juggie | rather then the port given to them in the sdp |
| 16:02.52 | CyberKnet | Juggie: you and me should write a sip client that follows the rules then ;) |
| 16:02.58 | CyberKnet | s/me/I/ |
| 16:03.25 | Juggie | i dont get it... |
| 16:03.34 | Juggie | maybe asterisk should be using the rtp range as the source ports as well |
| 16:04.05 | HA | is G.711 the same thing as ulaw? |
| 16:04.15 | CyberKnet | Juggie: it's an idea... but I dont know if it is accurate or not. |
| 16:05.02 | wasabi_ | okay here's some questions. |
| 16:05.33 | wasabi_ | these extensions, do they allow alpha chars? |
| 16:05.40 | wasabi_ | and if so, should we be using user names for our users? |
| 16:05.56 | wasabi_ | That way users can roam with their phones if neccassary. |
| 16:06.01 | wasabi_ | Or connect phones from home (vpn) |
| 16:06.15 | *** join/#asterisk Ridgeback (~Ridgeback@ppp218-189.lns1.adl2.internode.on.net) |
| 16:06.18 | Ridgeback | hello |
| 16:07.03 | AgiNamu | OK, any suggestions on what command name is best for an AGI function that sends audio to a TCP port? |
| 16:07.11 | AgiNamu | I dont want to use SEND AUDIO, cause its like SEND IMAGE or SEND TEXT |
| 16:07.19 | AgiNamu | but those are different things |
| 16:07.26 | AgiNamu | I was thinking FORWARD AUDIO |
| 16:07.28 | Ridgeback | AUDIO2TCP |
| 16:07.44 | AgiNamu | TCP AUDIO? |
| 16:07.53 | Ridgeback | sure |
| 16:07.56 | Ridgeback | sounds good |
| 16:08.05 | AgiNamu | yea |
| 16:08.11 | AgiNamu | thanks |
| 16:08.25 | Ridgeback | no prob :) |
| 16:08.35 | AgiNamu | TCP AUDIO <address> [<identifier>] and TCP STOP AUDIO |
| 16:08.36 | AgiNamu | cool |
| 16:08.49 | Ridgeback | hey does anyone know if nufone ever answer thier phones? |
| 16:08.58 | Ridgeback | AgiNamu, looks good |
| 16:09.23 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 16:09.27 | AgiNamu | nufone has phones? ;) |
| 16:09.27 | *** join/#asterisk jskcr|lappy (~jskcr@jskcr.user) |
| 16:09.34 | AgiNamu | bbl |
| 16:09.38 | *** join/#asterisk jeffreyeni (meltzer@exobit.exobit.org) |
| 16:09.42 | Ridgeback | AgiNamu, lol yeah! but they never answer them! |
| 16:11.32 | wasabi_ | So what about conference calls? |
| 16:11.37 | JerJer | happy friday the 13th |
| 16:11.52 | Qwell | JerJer: payday ;] |
| 16:12.03 | *** join/#asterisk tessier_ (~treed@222.253.65.236) |
| 16:12.44 | AgiNamu | SHIT! why did you have to bring that to my attention JerJer |
| 16:13.10 | AgiNamu | I wouldn't answer my phone on friday the 13th either |
| 16:13.14 | AgiNamu | you never know who it might be. |
| 16:13.18 | Sedorox | lol |
| 16:13.32 | HA | Friday the 13th always brings me good luck. |
| 16:13.36 | AgiNamu | I dont answer the phone after dark either. |
| 16:13.48 | AgiNamu | there's a lot of strange people out there. |
| 16:13.57 | wasabi_ | Okay we have this want, can asterisk do it reasonably: We want a conference call station in the conference room. |
| 16:14.00 | *** part/#asterisk darth-timeus (darth@200.105.128.61) |
| 16:14.06 | Ridgeback | wasabi_, conference calls? |
| 16:14.06 | wasabi_ | And we want everybody attending the meeting to be able to request transcripts. |
| 16:14.12 | AgiNamu | yes |
| 16:14.22 | AgiNamu | unless you mean TEXT transcripts |
| 16:14.24 | wasabi_ | no. |
| 16:14.29 | AgiNamu | in which case I'd say that asterisk is not an ASR system |
| 16:14.38 | wasabi_ | I understand. |
| 16:14.38 | wasabi_ | Voice. |
| 16:14.40 | AgiNamu | sure, but you need to write a program around the conferencing system |
| 16:14.51 | AgiNamu | because, AFAIK, it doesn't handle requesting transcripts. |
| 16:14.57 | AgiNamu | but it's certainly possible and not difficult. |
| 16:15.05 | AgiNamu | well, not too difficult. |
| 16:15.18 | AgiNamu | are you a C developer? |
| 16:15.30 | wasabi_ | I don't want to be. =) |
| 16:15.46 | AgiNamu | heh, that'll pass. :) |
| 16:15.47 | JerJer | AgiNamu: no we just never answer fones - we like the consent ringing noise around here |
| 16:15.52 | wasabi_ | Oh I see, the conference room itself gets an extension. |
| 16:15.55 | JerJer | constant |
| 16:15.57 | AgiNamu | JerJer, yea, it's soothing |
| 16:16.11 | AgiNamu | consentual ringing :) |
| 16:16.22 | AgiNamu | i think that's not allowed in some states. |
| 16:16.35 | wasabi_ | what is this ztdummy thing? |
| 16:16.41 | HA | i have never consented to being rung but that doesn't stop most people. |
| 16:16.41 | AgiNamu | wasabi_, recording it shouldn't be hard at all |
| 16:16.53 | AgiNamu | it's the "requesting a transcript" that falls into the realm of some management software. |
| 16:17.01 | wasabi_ | Yeah. |
| 16:17.04 | AgiNamu | which you could write in C, assembler, C++, C# |
| 16:17.12 | wasabi_ | Those in the meeting would have to somehow enter *, then their extension number. |
| 16:17.17 | wasabi_ | That can be programed though? |
| 16:17.20 | AgiNamu | sure |
| 16:17.34 | wasabi_ | So a bunch of dudes sitting around a table could type *myextension# or something |
| 16:17.36 | AgiNamu | you could make it only send transcripts on friday the 13th if you wanted to. |
| 16:17.40 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 16:17.41 | CyberKnet | AgiNamu: the perl folks are gonna get cross with you for neglecting them. And also the emacs folks =P" |
| 16:17.42 | wasabi_ | And I could write software to forward it to their mail box? |
| 16:17.55 | AgiNamu | CyberKnet, I only stuck to C languages :) |
| 16:18.02 | AgiNamu | wasabi_, right |
| 16:18.09 | CyberKnet | AgiNamu: Assembler is a C language now? =) |
| 16:18.09 | AgiNamu | again, not hard |
| 16:18.15 | AgiNamu | No, C is an assembler language |
| 16:18.15 | HA | or their email box. |
| 16:18.17 | wasabi_ | interesting. |
| 16:18.20 | AgiNamu | C is portable asm :P |
| 16:18.28 | wasabi_ | Okay, I think I have a more generic way of solving this. |
| 16:18.29 | CyberKnet | AgiNamu: heh |
| 16:18.40 | wasabi_ | Anybody anywhere on anycall at anytime should be able to hit a button and type an extension to send a transscript to. |
| 16:18.44 | newl | lda #$00 sta $d020 sta $d021 rts B) |
| 16:18.58 | AgiNamu | wasabi_, you could prolly get it done for like $400 or so |
| 16:19.02 | AgiNamu | if you hired someone |
| 16:19.09 | CyberKnet | newl: heh |
| 16:19.12 | AgiNamu | maybe a bit more. i dunno. never looked at the conferencing app code. |
| 16:19.18 | AgiNamu | but i can't imagine it'd be THAT difficult. |
| 16:19.45 | CyberKnet | ;)) |
| 16:19.51 | *** join/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 16:20.00 | AgiNamu | volunteering? yea for money |
| 16:20.09 | AgiNamu | I'm a great volunteer with some cash. |
| 16:20.15 | AgiNamu | s/with/for |
| 16:20.21 | CyberKnet | heh |
| 16:20.35 | CyberKnet | code for cash is not a bad arrangement. Keeps me comfortable. |
| 16:20.38 | AgiNamu | as soom as im finished workong on res_agi |
| 16:20.42 | flickerfly | I liked it better with with |
| 16:20.48 | AgiNamu | and getting rid of all ths retarded variable names |
| 16:21.02 | AgiNamu | and writing tcp audio forwarding |
| 16:21.08 | AgiNamu | yea, i'll look at conferencing. |
| 16:21.21 | CyberKnet | even better if the person paying lays in GPL/LGPL as part of the arrangement. |
| 16:22.01 | AgiNamu | yea, fine |
| 16:22.07 | AgiNamu | if they wanna do that to themselves, great. |
| 16:22.22 | AgiNamu | If they want to require that I delete it and forget I even did it, that's fine too. |
| 16:22.30 | AgiNamu | If they want to BSD it, that's even more fine. |
| 16:22.42 | AgiNamu | (like the PA168 people... yhea!) |
| 16:22.45 | *** join/#asterisk TheEmperor (TheEmperor@218.111.51.8) |
| 16:23.05 | AgiNamu | Centrality is a model for other hardware manufacturers |
| 16:23.25 | AgiNamu | They were so pleased that people contributed code to their firmware (like adding IAX2 native transfer) |
| 16:23.38 | AgiNamu | they decided to open up and BSD-license the entire codebase |
| 16:24.08 | AgiNamu | So it's the first commercially viable IAX2 device in existence. |
| 16:24.31 | AgiNamu | Surprising digium doesn't do something along that line |
| 16:24.39 | AgiNamu | like, make an IAX ATA and business class phone. |
| 16:29.10 | CyberKnet | AgiNamu: the IAXy isn't an IAX ATA? |
| 16:29.17 | *** join/#asterisk drumkilla (~russell@207.111.174.1) |
| 16:29.17 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 16:29.36 | jaiger | I've been having problems where my PSTN calls are periodically dropped, the console says "Hungup". can I find out which party hungup? |
| 16:30.59 | AgiNamu | CyberKnet, I meant a commercially viable one |
| 16:31.04 | Silik0n | BSD > GPL |
| 16:31.12 | AgiNamu | ULAW doesn't count as comprehensive codec support :P |
| 16:31.18 | Silik0n | hahahah |
| 16:31.28 | Silik0n | but uLaw has a higher MOS |
| 16:31.43 | AgiNamu | the PA168 supports ULAW/ALAW, G723, G729, GSM, soon iLBC and eventually Speex. |
| 16:31.46 | Silik0n | so its obviously the best codec to use |
| 16:32.22 | AgiNamu | heh |
| 16:32.25 | AgiNamu | well, im outta here |
| 16:32.26 | AgiNamu | cya |
| 16:34.46 | CyberKnet | later |
| 16:34.55 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 16:34.55 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 16:35.09 | atporter | anyone got an iax<->pstn provider they're happy with? |
| 16:35.12 | CyberKnet | early morning, eh? |
| 16:35.23 | CyberKnet | atporter: outbound, or DID? |
| 16:35.27 | atporter | both |
| 16:35.42 | CyberKnet | I'm only outbound with voipjet right now. |
| 16:35.43 | atporter | possibly different providers though, I guess |
| 16:35.51 | CyberKnet | reasonably happy with the little use I have. |
| 16:35.57 | atporter | I'm not too happy with sixtel |
| 16:36.11 | CyberKnet | atporter: For what reasons? |
| 16:37.10 | atporter | had some spotty outbound calls, close to two weeks since I ordered a DID and I've not gotten it, over 36 hours since I opened a trouble ticket asking why and it's not been touched |
| 16:38.08 | *** join/#asterisk Inv_arp (junya@adsl-3-247-188.mia.bellsouth.net) |
| 16:38.08 | atporter | voipjet.com looks pretty nice |
| 16:38.09 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 16:39.24 | Silik0n | atporter: i have asterlink and I am happy with them for inbound 800 and for outbounds 9they just dont have a many NPAs so a local did you might not find from here |
| 16:39.24 | blitzrage | atporter: mixnetworks.com |
| 16:39.28 | Silik0n | hahah |
| 16:39.31 | Silik0n | mix |
| 16:39.35 | JerJer | bell.ca |
| 16:39.40 | blitzrage | lol |
| 16:39.54 | blitzrage | JerJer: prices are probably going up for bell.ca now that they are going to be regulated by the CRTC |
| 16:42.07 | cpatry | blitzrage: whatcha think about all that? |
| 16:42.29 | blitzrage | cpatry: probably good for me ;) |
| 16:42.45 | cpatry | it offers a great opportunity for videotron |
| 16:42.51 | bjohnson | bell.ca? what did I miss? |
| 16:42.52 | blitzrage | uhhh, yah :) |
| 16:43.15 | blitzrage | bjohnson: CRTC is regulating the major phone companies VoIP offerings so they can't use VoIP as a loss leader |
| 16:43.19 | bjohnson | what bell product is being discussed? |
| 16:43.26 | cpatry | their voip services started close to montreal, heard their services are pretty good. |
| 16:43.33 | bjohnson | oh .. bell had prices for voip? |
| 16:43.44 | cpatry | bjohnson: nope, not released yet. |
| 16:44.20 | cpatry | i hope its gonna be better then their sympatico (HSE) for stability :) |
| 16:45.01 | blitzrage | I would fricken hope so |
| 16:45.12 | blitzrage | sympatico sucks in my experience though |
| 16:45.24 | bjohnson | don't know .. hacen't used them in years |
| 16:45.42 | cpatry | sympatico's crap here. |
| 16:45.42 | Juggie | ahhh, i finally see the SIP nat problem |
| 16:46.00 | Juggie | some clients like to use the rtp source port to return their RTP packets |
| 16:46.08 | Juggie | rather then the port in the SDP |
| 16:46.36 | bjohnson | my philosophy is that they have been dropping prices since competition has been allowed. Therefore, they were screwing us before. Therefore even if they meet the lowest price, I won't be using them since I WANT their competition to be around for a long time so they can't screw me again |
| 16:46.47 | Juggie | i wish i could control the source port range for RTP from asterisk. |
| 16:46.54 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
| 16:48.02 | bjohnson | generally my philosophy against any dominant leader in any market |
| 16:48.16 | Juggie | rogers is waiting for bell to make a move in ontario |
| 16:48.22 | Juggie | rather, bell is waiting for rogers |
| 16:48.34 | Juggie | bell doesnt want to step on its pstn service any earlier then it has to |
| 16:48.48 | Juggie | which i think is a mistake because if rogers comes out with a killer solid service, they will get the customers |
| 16:49.06 | Juggie | bell should have moved first to force rogers to come out with a service that wasnt 100% ready |
| 16:49.24 | blitzrage | Rogers Internet sucks bawlz |
| 16:49.29 | blitzrage | but good cell service |
| 16:49.30 | cpatry | wow |
| 16:49.31 | cpatry | http://www.physorg.com/news4055.html |
| 16:49.34 | Juggie | its already going to be super hot |
| 16:49.45 | Juggie | rogers isnt using ethernet for voip |
| 16:49.56 | Juggie | but instead an ata which goes right onto the coax |
| 16:50.02 | Gand_DJ | MTS Voip is being targetted at businesses, and not residential |
| 16:50.22 | Juggie | it is going to plug into the wall, and have a built in battery |
| 16:50.29 | Juggie | so if the power goes out, you keep your phone servicve |
| 16:50.33 | Juggie | *service |
| 16:52.45 | blitzrage | Juggie: yep, probably requirements for 911 |
| 16:52.46 | Dishwasha | I have the eyeBeam SIP softphone from Xten and for some reason it is telling me "Network does not support park" even though I have both a features.conf and a parking.conf and have included the parkedcalls context in my extensions.conf |
| 16:52.48 | Dishwasha | any ideas? |
| 16:52.50 | harryvv | juggie, yea cable is going to take a majority of telus phone traffic away from them. |
| 16:53.03 | harryvv | shaw cable and rogers as well as others. |
| 16:53.05 | blitzrage | Cogeco probably has the best network I've seen so far |
| 16:53.07 | Juggie | rogers may be an asshole company |
| 16:53.11 | Juggie | but their network is pretty solid |
| 16:53.18 | Juggie | my internet never ever ever goes down |
| 16:53.18 | blitzrage | I get a true 10mbit/1mbit pipe for $60/mth |
| 16:53.34 | blitzrage | mines gone down like... twice I think for a total down period of like 20 mins |
| 16:53.48 | harryvv | in what time range |
| 16:54.00 | JerJer | we get gige for $100 a month - true gige both ways :P |
| 16:54.11 | Juggie | yah right |
| 16:54.20 | blitzrage | JerJer: oooooooo :) |
| 16:54.30 | jontow | take me to your uplink |
| 16:54.31 | jontow | etc |
| 16:54.32 | jontow | :) |
| 16:54.35 | blitzrage | harryvv: like... 2 years maybe |
| 16:54.37 | Juggie | i can get E10 for 700$ a month |
| 16:54.40 | Juggie | un metered |
| 16:54.50 | blitzrage | Juggie: I'd never by something that was metered |
| 16:54.55 | blitzrage | I don't believe in it :) |
| 16:55.02 | Juggie | hence, un-metered :) |
| 16:55.09 | Juggie | 10mbit duplex 700$ |
| 16:55.33 | blitzrage | ugh... too much |
| 16:55.46 | blitzrage | $60 10mbit/1mbit unmetered |
| 16:56.15 | *** join/#asterisk Nemesis760 (~nemesis@63-227-245-38.bois.qwest.net) |
| 16:56.18 | bjohnson | harryvv: per day |
| 16:56.27 | blitzrage | if I got 10 connections it'd still be only $600 :) |
| 16:56.31 | Nemesis760 | Hello all. |
| 16:57.07 | harryvv | bjohnson: I was replying to blitzrage comment on down time. |
| 16:57.08 | harryvv | :) |
| 16:57.12 | Juggie | blitzrage, i understand its 60$ |
| 16:57.19 | Juggie | but you understand that E10 is not cable modem |
| 16:57.23 | Juggie | its ethernet over fiber |
| 16:57.27 | Juggie | its full 10mbit outbound |
| 16:57.33 | Juggie | and its a business line |
| 16:57.49 | Nemesis760 | I need a bunch of Voice PRIs, and got a quote of $357/ea from TimeWarner... these are 0-mile, we'd be colo'd in their facility. Does this seem reasonable? |
| 16:58.01 | bjohnson | yes |
| 16:58.14 | Juggie | 357 aint bad |
| 16:58.17 | mutilator | a bunch = ? |
| 16:58.29 | bjohnson | 4 |
| 16:58.34 | Nemesis760 | 10 to start... could be 50+ within 6 months. |
| 16:58.43 | mutilator | ehm |
| 16:58.45 | bjohnson | one = 1, couple = 2, few =3, bunch =4 |
| 16:58.56 | harryvv | dont be to suprised if cable companies decide to mount ether/pstn/cable wall mount boxes in the near future |
| 16:59.01 | blitzrage | anyone notice the new xbox looks like a Dell? :) |
| 16:59.33 | *** join/#asterisk easimon (~easimon@localhorst.kawo2.RWTH-Aachen.DE) |
| 16:59.33 | Nemesis760 | They said since it's 0-mile anyways, wouldnt save anything by laying over a DS3. |
| 17:00.32 | AgiNamu | blitzrage, but Dells don't have triple-core PowerPCs at 3.2GHz |
| 17:01.06 | CyberKnet | yessir they do not. |
| 17:01.15 | AgiNamu | anyways, consoles suck |
| 17:01.22 | Nemesis760 | Anyone here have success stories with more than 2 Quad-Span cards in comodity hardware (IE. Dell Dual XEON) |
| 17:01.28 | AgiNamu | by the time the xbox2 ships, nVidia will be kicking their ass |
| 17:01.40 | CyberKnet | AgiNamu: by Q3? |
| 17:01.59 | AgiNamu | yea, i wouldn't be surprised. |
| 17:02.06 | Nemesis760 | I'm working on the assumption that 2 is max. ?? |
| 17:02.13 | CyberKnet | AgiNamu: CPU-wise x86 will only be dual core by then. |
| 17:02.24 | AgiNamu | CyberKnet, no, i meant as far as the graphics go |
| 17:02.26 | blitzrage | infinite core! |
| 17:02.27 | CyberKnet | AgiNamu: I've never much been one for ATI gpus anyway |
| 17:02.39 | AgiNamu | and you can put multiple dual cores on a single mobo |
| 17:02.48 | blitzrage | I'm not a gamer, so it just needs to be able to do 1600x1200 at 32bit :) |
| 17:02.50 | AgiNamu | so I think having a quad-core system wont be an issue |
| 17:02.56 | blitzrage | and 85 Hz refresh |
| 17:03.07 | AgiNamu | I hate CRT |
| 17:03.09 | CyberKnet | AgiNamu: you can already buy a quad machine. |
| 17:03.11 | blitzrage | I love CRT |
| 17:03.14 | blitzrage | I hate LCD :) |
| 17:03.15 | AgiNamu | CyberKnet, yep. |
| 17:03.21 | CyberKnet | AgiNamu: 2xdual core does not a quad core make |
| 17:03.23 | AgiNamu | CRTs make me sick |
| 17:03.27 | AgiNamu | sure it does |
| 17:03.32 | blitzrage | really? I have that problem with LCD |
| 17:03.34 | AgiNamu | there are four cores :P |
| 17:03.35 | CyberKnet | no, it makes two dual core cpus |
| 17:03.43 | CyberKnet | it doesn't make a quad core cpu |
| 17:03.49 | AgiNamu | no of course not |
| 17:04.13 | *** join/#asterisk iamnotbob (~nolan@216.199.159.79) |
| 17:04.21 | AgiNamu | course, i wonder what the xbox price will be |
| 17:04.27 | CyberKnet | now, it might be that because there are *four* cores instead of three that it might run faster, but if you disabled one of those cores I very much doubt it would run anywhere near comparison to a three core PowerPC processor |
| 17:04.33 | AgiNamu | anyways, it doesnt matter. Xbox will still suck cause no one writes good games for it. |
| 17:04.36 | CyberKnet | AgiNamu: I wouldn't be surprised to see it in the 299-350 range |
| 17:04.44 | AgiNamu | so unless you want to play Halo and "Some dumbass super Sport" |
| 17:04.50 | CyberKnet | Halo2 is amazingly good =) |
| 17:04.51 | AgiNamu | you're screwed. |
| 17:04.58 | AgiNamu | MS goes on an on about how good Halo is |
| 17:05.02 | AgiNamu | like, it sold a few million copies. |
| 17:05.03 | CyberKnet | dumbass super sport is and always will be dumbass super sport |
| 17:05.10 | AgiNamu | like that's some kind of success |
| 17:05.24 | Sedorox | Halo is only good for Red Vs. Blue |
| 17:05.30 | CyberKnet | I believe the success was that it was multi-milion copies on launch day |
| 17:05.55 | AgiNamu | yea, all those poor guys with Xbox and nothing to play |
| 17:05.56 | iamnotbob | does anyone know how to get asterisk to play moh (under fedora 3) for sip devices |
| 17:06.02 | blitzrage | AgiNamu: whaaat?! |
| 17:06.02 | pussfeller | didnt ms hire rare to make halo |
| 17:06.15 | blitzrage | AgiNamu: I prefer xbox over that playstation bullshit |
| 17:06.31 | AgiNamu | Well, as soon as Microsoft gets Squaresoft on board |
| 17:06.32 | pussfeller | i know they bought rare, but I don't know what they made for them |
| 17:06.48 | blitzrage | haha... look what I've started, an xbox discussion :) |
| 17:06.57 | Fanguin | Hello, can somebody help me and give me a hint why this six lines long agi php script does not play a file? http://pastebin.ca/11671 |
| 17:07.10 | AgiNamu | Final Fantasy X and X-2 have sold 10 million copies alone. |
| 17:07.18 | AgiNamu | Let alone the millions and millions from the rest of the series. |
| 17:07.27 | AgiNamu | Once MS understand taht you need good games, things might go o.k. |
| 17:07.32 | AgiNamu | but their current titles suck hard. |
| 17:07.48 | AgiNamu | and Halo on Xbox.... wtf.... |
| 17:08.08 | iamnotbob | xbox & like is a little off topic for this channel |
| 17:08.23 | AgiNamu | iamnotbob, no it's not. we're discussing how chanspy can work on xbox. |
| 17:08.50 | AgiNamu | meh, xbox only does 1080i |
| 17:08.53 | AgiNamu | what kind of lame crap is that |
| 17:09.21 | Juggie | 1080p is still expensive |
| 17:09.42 | Dishwasha | I have the eyeBeam SIP softphone from Xten and for some reason it is telling me "Network does not support park" even though I have both a features.conf and a parking.conf and have included the parkedcalls context in my extensions.conf |
| 17:09.47 | *** join/#asterisk t3chno (~endeavor@69.158.62.219) |
| 17:09.54 | AgiNamu | dont the dell lcd tvs support it |
| 17:10.19 | iamnotbob | ok......sure... hdtv is still a bit off... |
| 17:10.42 | AgiNamu | no its not. I've got a 1280x1024 display that im using right now for xchat |
| 17:11.30 | t3chno | hey, i got a real n00b question for this.. can i get voip like vonage or something.. and run it to asterisks so it answers my vonage line, and then make it transfer any calls to a phone plugged into the modem on the box? |
| 17:11.56 | Qwell | t3chno: vonage is pretty locked down |
| 17:12.06 | t3chno | oh really.. hmm |
| 17:12.19 | iamnotbob | yes * will work with vonage... don't ask me how to do it. |
| 17:12.28 | t3chno | ok |
| 17:12.38 | t3chno | but can i have it transfer to a phone plugged into the modem? |
| 17:12.49 | t3chno | like "press 0 for assistance" and itll ring a phone |
| 17:13.38 | Dishwasha | According to http://support.xten.com/viewtopic.php?t=3013&highlight=park&sid=a8720beed48c4d34296c9627a606af57 Asterisk dosn't properly support the parked call feature |
| 17:13.39 | iamnotbob | basicaly, but the "modem" should be already functioning as an interface.... |
| 17:14.12 | t3chno | ok im really new to this |
| 17:14.15 | t3chno | what does that mnea |
| 17:14.18 | t3chno | *mean |
| 17:14.26 | iamnotbob | I have read somewhere that * is not fully sip compliant, and from what little I know of it, that is a true statement. |
| 17:14.34 | iamnotbob | * = asterisk |
| 17:14.37 | t3chno | yeah |
| 17:14.39 | AgiNamu | is anything fully sip compliant? |
| 17:14.44 | blitzrage | what IS fully SIP compliant? |
| 17:14.45 | bjohnson | t3chno: the "modem" would have to be a fxs or fxo .. not a modem |
| 17:14.49 | Dishwasha | sipexchange from pingtel should be |
| 17:14.53 | blitzrage | I read the SIP RFC... I don't think even that is SIP complient :) |
| 17:14.59 | t3chno | ok, what is fxs or fxo |
| 17:15.05 | blitzrage | Dishwasha: and i bet its not 100% |
| 17:15.06 | Qwell | ~fxofxs |
| 17:15.09 | jbot | it has been said that fxofxs is An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
| 17:15.27 | Dishwasha | didn't pingtel help develop the RFCs? |
| 17:15.34 | bjohnson | and vonage will work with * if you pay for subscru=iption, pay for soft phne service, and kiss away any minutes of the main service |
| 17:15.43 | bjohnson | there are much better options |
| 17:15.45 | iamnotbob | yes there are things fully sip compliant, but still thete are deviations.. |
| 17:15.45 | blitzrage | I don't remember seeing that... but who knows |
| 17:15.49 | Qwell | I thought * wasn't allowed, even with the softphone account? |
| 17:16.01 | iamnotbob | freeworld dialup, and iconnecthere are some others |
| 17:16.02 | blitzrage | fuck vonage(tm) |
| 17:16.04 | Dishwasha | That's what they said during a sales meeting we had where they were a partner |
| 17:16.16 | t3chno | hmm |
| 17:16.42 | t3chno | hang on, what if i just plugged a vonage hardware modem into the line in on a modem on the * box? |
| 17:16.53 | t3chno | wouldnt it just assume its a normal phone line? |
| 17:16.54 | AgiNamu | t3chno, forget modem |
| 17:17.02 | AgiNamu | but yes, you can take your adapter, put that analog to Asterisk |
| 17:17.08 | Dishwasha | Of course, Microsoft could claim the same thing for the XML working group but they came out with their own incompliant XML |
| 17:17.13 | t3chno | right ok |
| 17:17.14 | AgiNamu | Dishwasha, that makes no sense |
| 17:17.18 | blitzrage | I don't understand why you would pay so much for vonage and go through all sorts of hoops trying to get it into Asterisk, just use a more open service provider for less money |
| 17:17.28 | Dishwasha | AgiNamu: What doesn't? |
| 17:17.38 | AgiNamu | their own incompliant XML |
| 17:17.51 | Dishwasha | Yes, Microsoft XML is not complain with true XML standards |
| 17:17.55 | Dishwasha | compliant |
| 17:18.05 | bjohnson | t3chno: forget modems |
| 17:18.13 | t3chno | ok well, what about the call transfer thing? give the ability to transfer the call to a phone plugged into the * box |
| 17:18.27 | Dishwasha | true XML standard = W3C |
| 17:18.27 | bjohnson | you could plug a vonage ata into an fxs .. but then you're buying double the hardware |
| 17:18.31 | AgiNamu | Dishwasha, what is "microsoft XML?" |
| 17:18.35 | iamnotbob | use a ata adapter rather than a modem or asterisk's compliant hardware |
| 17:18.45 | AgiNamu | and what is it not compaint with |
| 17:18.52 | blitzrage | I'm blitzrage SIP compliant |
| 17:19.04 | AgiNamu | specifics man |
| 17:19.05 | Dishwasha | It encompasses a lot of things such as SOAP |
| 17:19.07 | blitzrage | :D |
| 17:19.17 | AgiNamu | so you're saying MS soap doesn't work on other platforms? |
| 17:19.21 | Dishwasha | And like I said, the W3C group |
| 17:19.26 | Dishwasha | AgiNamu: No, you said that |
| 17:19.33 | AgiNamu | yes, you did. but you didnt say what its incompatible with |
| 17:19.42 | *** join/#asterisk jdg (~jdg@CA03F809.adsl.mana.pf) |
| 17:19.42 | AgiNamu | its like idiots who say "IE should be more standard compliant!" |
| 17:19.43 | Dishwasha | working on other platforms != standards compliant |
| 17:19.44 | bjohnson | t3chno: what you are describing will work .. it is very basic. Just replace "vonage" with "decent voip provider" and "modem" with "fxs" or "fxo" |
| 17:20.23 | t3chno | haha ok.. i wasnt really looking into voip and not specifically vonage |
| 17:20.24 | AgiNamu | oh, so this is a theoretical "it's not w3c compliant" thing |
| 17:20.24 | t3chno | but ok |
| 17:20.30 | bjohnson | AgiNamu: right .. it's the idiots who design ONLY for IE |
| 17:20.32 | AgiNamu | yea, I'm fine with that |
| 17:20.32 | t3chno | but what is fxs/fxo? |
| 17:20.43 | Dishwasha | AgiNamu: There's this thing called the internet, look it up and you find lots of information on it |
| 17:20.45 | AgiNamu | or the people who ask for IE to support CSS3 |
| 17:20.46 | bjohnson | t3chno: hardware that you buy |
| 17:21.02 | Qwell | how about supporting css at all properly first? |
| 17:21.12 | t3chno | ok.. and the difference between that and a modem? |
| 17:21.13 | AgiNamu | Dishwasha, that's exactly what I thought. nonsense "it's not compliant" claims that dont really say anything. |
| 17:21.20 | t3chno | sorry im really new to the whole telephony thing |
| 17:21.23 | bjohnson | how about sites not catering to IE and designing standards compliant sites |
| 17:21.36 | bjohnson | t3chno: 1. ability 2. cost |
| 17:21.39 | AgiNamu | bjohnson, IE is quite standards compliant if you tell it to. |
| 17:21.44 | Dishwasha | Because we've deviated from my using microsoft XML as an example of a vendor who had a seat in the standards process and created something non-compliant to compare to pingtel having a seat in the SIP standards process and potentially making a SIP PBX that is not SIP compliant |
| 17:21.55 | AgiNamu | but to maintain backword compatability, it still has even the netscape emulation hacks inside it |
| 17:22.00 | bjohnson | AgiNamu: mine won't listen to me |
| 17:22.05 | t3chno | um ok |
| 17:22.06 | t3chno | hm |
| 17:22.07 | AgiNamu | Dishwasha, you're right. |
| 17:22.51 | t3chno | ok so fxs/fxo is like a supercharged modem? |
| 17:22.55 | Qwell | ~fxofxs |
| 17:22.56 | jbot | somebody said fxofxs was An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
| 17:22.56 | bjohnson | nope |
| 17:23.15 | bjohnson | they are voip hardwarea modem is not voip hardware |
| 17:23.21 | Dishwasha | so, full circle, does anybody know how to get the parked call feature working with Xten's eyephone? |
| 17:23.36 | t3chno | oh ok |
| 17:23.42 | t3chno | hmm |
| 17:23.53 | jets | What is everyone's favorite linux soft phone |
| 17:23.55 | AgiNamu | t3chno, really, go read some documentation and general telephony info |
| 17:24.07 | t3chno | yeah |
| 17:24.08 | AgiNamu | jets, Wine + Firefly? |
| 17:24.12 | t3chno | good idea |
| 17:24.13 | t3chno | haha |
| 17:24.37 | bjohnson | you can run voip over a IP network created by a modem (but very good quality) .. but the modem is just acting like a LAN switch in that case .. not voip hardware |
| 17:24.59 | jets | Firefly works well under wine? cool. |
| 17:25.12 | Dishwasha | you could try sjphone jets |
| 17:25.16 | bjohnson | ie .. you can't plug a phone into it and hear something other than beeps and whistles |
| 17:25.17 | Dishwasha | not a favorite but it works |
| 17:25.40 | Dishwasha | I thought the knew gnome had a sip phone built in |
| 17:25.46 | Dishwasha | haven't confirmed tho |
| 17:25.48 | AgiNamu | jets, no i have no clue |
| 17:25.50 | AgiNamu | i was makin a joke |
| 17:25.56 | ronn | is there a way to change the codec of a sip channel on the fly? just before dialing the channel ? instead of reading from sip.conf? |
| 17:26.06 | bjohnson | iaxcomm is a linux softphone |
| 17:26.17 | bjohnson | I've also used linphone. kphone is popular |
| 17:26.18 | AgiNamu | ronn yes |
| 17:26.23 | bjohnson | most people end up buying hardware |
| 17:26.29 | AgiNamu | but i dont know if it's available in CVS |
| 17:26.33 | AgiNamu | I wrote a patch to do exactly that |
| 17:26.37 | AgiNamu | lemme find it |
| 17:26.39 | ronn | AgiNamu: how do yo udo that? is there an application for that |
| 17:26.47 | AgiNamu | no, its a channel driver patch |
| 17:26.52 | ronn | oh i see ok |
| 17:26.53 | AgiNamu | Oh wait. for sip, i didnt write it. i dont think |
| 17:26.57 | jets | I have a hard phone just want something for travelling i spose |
| 17:27.00 | AgiNamu | there is a SIP_CODEC variable you can try |
| 17:27.02 | jets | thanks gents |
| 17:27.15 | CyberKnet | bjohnson: don't we all = |
| 17:27.17 | CyberKnet | =) |
| 17:27.37 | ronn | SIP_CODEC: in which source file is that? |
| 17:27.37 | jets | I have three pri's to play with but they aren't too happy today |
| 17:27.44 | CyberKnet | crying shame to see six of them here at work every day with onlu 80% utilization =) |
| 17:27.47 | AgiNamu | http://bugs.digium.com/view.php?id=3346 |
| 17:28.06 | AgiNamu | ronn, that link has my patch |
| 17:28.13 | ronn | AgiNamu: got that. thanks |
| 17:28.14 | AgiNamu | adds a CODEC_OVERRIDE channel var |
| 17:28.16 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 17:28.23 | AgiNamu | the SIP_CODEC didnt do shit for me |
| 17:28.26 | AgiNamu | as it's for something else |
| 17:28.28 | AgiNamu | not outgoing calls |
| 17:29.05 | ronn | i need it for outgoing calls |
| 17:29.46 | AgiNamu | yea, mine works for outgoing |
| 17:29.56 | ronn | ok. |
| 17:30.09 | AgiNamu | although, i dont know if it works on chan_sip. i think so |
| 17:30.10 | AgiNamu | if not, ping me |
| 17:30.25 | ronn | yes. i will |
| 17:31.11 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 17:31.15 | ManxPower | What time is it in Europe at the moment, and do they have daylights saving time? |
| 17:31.27 | ManxPower | Specifically Sweeden, Frande, Benelux, etc |
| 17:31.37 | ronn | it's 7.30 CET |
| 17:32.03 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 17:32.04 | ManxPower | ronn: Thanks. |
| 17:32.16 | ManxPower | That's what I thought. |
| 17:32.37 | ManxPower | 7 hours ahead of Central USA time |
| 17:34.29 | Deryl | (for those not monitoring the freebsd sec list) |
| 17:34.31 | Deryl | HTT Vuln Info Updated: ftp://ftp.freebsd.org/pub/FreeBSD/CERT/advisories/FreeBSD-SA-05:09.htt.asc |
| 17:34.39 | Deryl | this is NOT limited to freebsd. |
| 17:34.53 | *** join/#asterisk RaYmAn-Bx (rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 17:35.03 | *** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 17:35.06 | Qwell | Deryl: Thats what you get for using Intel. :P |
| 17:36.02 | Deryl | Qwell: waaa |
| 17:38.25 | *** join/#asterisk CyberGlitch2 (~jsievert@12.150.244.162) |
| 17:38.42 | AgiNamu | Deryl, is that freebsd specific |
| 17:38.50 | Deryl | no |
| 17:38.59 | Deryl | i just said it wasn't |
| 17:39.10 | CyberGlitch2 | hello all |
| 17:39.17 | AgiNamu | duh, yes you did |
| 17:39.34 | Deryl | hehe |
| 17:39.51 | *** join/#asterisk rcam (~rcammobil@adsl-218-151-77.jax.bellsouth.net) |
| 17:40.19 | rcam | Hello. |
| 17:40.37 | *** join/#asterisk cmk (~cmk_@p54A3CFAE.dip.t-dialin.net) |
| 17:41.25 | durex | heheh i love this: cvsup /etc/supfile-stable && cd /usr/src && make buildworld && make installworld && reboot |
| 17:41.27 | durex | ;-) |
| 17:41.47 | Deryl | don't forget mergemaster |
| 17:42.13 | durex | yes yes hiaehaih and to compile the kernel too! ;-) |
| 17:42.13 | Deryl | especially so you have the most current cvs output tags in files that have them |
| 17:42.24 | durex | deryl, yes sure, relax about it... but is it what I like in BSDs... specially fbsd |
| 17:43.50 | Deryl | durex: why do i need to relax? i'm not uptight. |
| 17:44.09 | wasabi_ | So what about integrating of asterisk into other directory platforms? Any work been done? |
| 17:44.13 | wasabi_ | Specifically LDAP. |
| 17:46.30 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 17:47.25 | CyberGlitch2 | so i just got my first asterisk@home box set up and i was wondering what other cool stuff you could do with it? |
| 17:48.02 | jontow | if you can find the soundfiles you can make it squeal like a pig |
| 17:48.10 | CyberGlitch2 | weeeeee |
| 17:48.19 | bjohnson | wasabi_: you can do ldap lookups |
| 17:49.22 | CyberGlitch2 | how do i go about setting up confrence rooms? |
| 17:49.34 | bjohnson | wasabi_: there has been talk about everything from making a read/write ldap system for voicemail, disa authentication, followme scripts and even full storage systems for an entire ldap based config system |
| 17:49.43 | Nugget | CyberGlitch2: reading the documentation would be an excellent start. |
| 17:49.44 | bjohnson | I don't thnik much has been done with that |
| 17:50.01 | Silik0n | .last -msg |
| 17:50.06 | bjohnson | CyberGlitch2: the default system is called meetme |
| 17:50.48 | CyberGlitch2 | Nugget, i suppose but i was hoping that you might bless with with you wisdom and insite on where a good guide is |
| 17:51.00 | *** join/#asterisk ChkDigit (~mike@static65-87-228-18.regina.accesscomm.ca) |
| 17:51.23 | bjohnson | the wiki is always a good place to start |
| 17:51.36 | bjohnson | or the asterisk command line with "show application" |
| 17:52.16 | Juggie | does anyone possibly not have a gmail account |
| 17:52.22 | Juggie | i have like 50 invites here, if anyone wants |
| 17:56.21 | *** join/#asterisk HerVegas (~a@216.84.62.226) |
| 17:59.20 | durex | Juggie lol |
| 18:00.50 | Grooby | so who here uses voipjet? |
| 18:01.14 | harryvv | i have it but yet to use it. now i have a firewall that can pass sip traffic. |
| 18:01.32 | Inv_arp | Grooby: i do (outbound only) works fine |
| 18:01.51 | Grooby | do they provide inbound calls? |
| 18:02.27 | Inv_arp | Grooby: not sure... i 'am going to use iax.xx for inbound |
| 18:02.36 | Inv_arp | err iax.cc |
| 18:03.08 | harryvv | I have had lots of good luck with iax.cc |
| 18:04.03 | Inv_arp | harryvv: been using BV but to many probs timeouts, only ulaw support etc... |
| 18:04.21 | Inv_arp | so its gonna be either VP connect or iax.cc |
| 18:04.25 | Grooby | i am using BV |
| 18:04.41 | *** join/#asterisk coppice (~chatzilla@43.198.17.210.dyn.pacific.net.hk) |
| 18:04.43 | Grooby | not really happy w/ their outtage at this point.... |
| 18:05.27 | Inv_arp | Grooby: me too, i dont think its working now |
| 18:05.58 | Grooby | I currently got a backup for outgoing calls now |
| 18:06.23 | Grooby | but still tryiing to find something better at least w/ some sort of unlimited domestic calls |
| 18:06.34 | Grooby | (that is also stable) |
| 18:07.00 | HerVegas | Grooby,Inv_arp: do you know what kind of outage iax.cc has? |
| 18:07.26 | Inv_arp | HerVegas: not with them yet... but i hear there pretty good |
| 18:07.38 | Grooby | i dunno..never used them...just heard of them |
| 18:07.39 | Grooby | hehehe |
| 18:08.06 | Inv_arp | darwin35: jobs like that exist? nice |
| 18:08.09 | HerVegas | thanks, I was just checking |
| 18:08.12 | darwin35 | now to pack and move |
| 18:09.12 | harryvv | what is BV inv? |
| 18:09.19 | harryvv | broadvoice? |
| 18:09.52 | harryvv | Her, its not often. |
| 18:10.00 | Inv_arp | harryvv: yeap |
| 18:10.17 | *** join/#asterisk channan (~channan99@66.180.121.185) |
| 18:10.18 | harryvv | in fact, I dont know if asterisk records voip service time outs |
| 18:10.19 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 18:11.27 | *** join/#asterisk Flyboy6440 (~Bobo@192.76.82.90) |
| 18:11.40 | Flyboy6440 | ok have an odd question i suppose.. |
| 18:11.44 | shido6 | ask |
| 18:12.00 | Flyboy6440 | zap channels support *60.. this works great for adding the last caller to the blacklist.. |
| 18:12.07 | *** part/#asterisk jdg (~jdg@CA03F809.adsl.mana.pf) |
| 18:12.11 | Flyboy6440 | however iaxy devices do not support this... |
| 18:12.13 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
| 18:12.16 | Dishwasha | Alright, looks like the ability to do park calls is a transfer feature, so it just so happens that the specific park button on the eyePhone doesn't work is all |
| 18:12.25 | Flyboy6440 | anyway to add this type of functionality? |
| 18:12.54 | shido6 | what do you mean Flyboy6440 |
| 18:12.58 | shido6 | stick it in the dialplan |
| 18:13.01 | Flyboy6440 | yea |
| 18:13.04 | shido6 | the iaxy isnt as smart as you think it is |
| 18:13.05 | ManxPower | Flyboy6440: You can do it yourself or pay a bounty for the feature. |
| 18:13.13 | Dishwasha | the odd thing is I don't hear the park announcement like I'm supposed to |
| 18:13.31 | shido6 | Dishwasha, you watch "Dragon" one time too many? |
| 18:13.44 | ManxPower | Dishwasha: hearing the park announcement requires Supervised/Consultative transfer. Sounds like your phone is only doing Blind transfer. |
| 18:13.44 | Dishwasha | shido6: YES!!! You're like the only person in the world who gets it! |
| 18:13.54 | shido6 | thats one of my favorite parts |
| 18:14.05 | shido6 | and I remind myself of it everytime I slack off |
| 18:14.09 | Flyboy6440 | was thinking of sticking it in the dialplan, but was'nt sure how i suppose |
| 18:14.12 | ManxPower | Flyboy6440: The IAXy sends each digit AS IT'S DIALED to Astrisk, so you can handle it in the dialplan like Shido said |
| 18:14.16 | Dishwasha | ManxPower: How would I tell if I can do supervised or consultative transfer? |
| 18:14.30 | Dishwasha | shido6: Yeah, love that movie. |
| 18:14.47 | ManxPower | Dishwasha: Supervised/Consultative transfer allows you to talk to the dialed party before completing the transfer. Check the docs for your device. |
| 18:15.04 | Dishwasha | okay, so that's a SIP phone feature then, thanks |
| 18:15.20 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 18:15.21 | ManxPower | Dishwasha: EVERYTHNING is a device feature in SIP |
| 18:16.11 | wasabi_ | two phones on a lan with asterisk as a proxy, do those phones talk directly to each other or thru asterisk? |
| 18:16.38 | harryvv | asterisk is the bridge that makes it possible. |
| 18:16.59 | wasabi_ | oh, so two phones can't call each other, like two software phones can |
| 18:18.03 | *** part/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 18:19.22 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
| 18:19.41 | bjohnson | dragon is the one with sean connery as the dragon's voice? |
| 18:20.10 | wasabi_ | dragonheart. |
| 18:20.14 | Cresl1n | yeah |
| 18:20.18 | wasabi_ | ? |
| 18:20.21 | Dishwasha | ManxPower: Looks like those features are a CVS feature, not a 1.0.X feature |
| 18:20.24 | harryvv | yes |
| 18:21.27 | bjohnson | what's "Dragon" then? |
| 18:21.37 | Dishwasha | The Bruce Lee Story |
| 18:21.45 | *** join/#asterisk Luke-Jr (~luke-jr@207.192.221.172) |
| 18:21.56 | Dishwasha | Staring Brandon Lee |
| 18:22.33 | ManxPower | Dishwasha: That's just the # transfter hack. Most people don't need it |
| 18:22.49 | bkw_ | actually valetparking can do what he wants |
| 18:22.57 | harryvv | I think dialpad was one of the first to offer telephony over the net before cisco had voip approved? |
| 18:23.09 | darwin35 | I need valetparking bkw will you park for me |
| 18:23.16 | Dishwasha | ManxPower: ah, well realistically as long as I can do a blind, my customers are doing overhead paging anyway when they park so I' |
| 18:23.19 | Dishwasha | ll be alright |
| 18:23.22 | bkw_ | http://www.pbxclue.com/asterisk_apps/ |
| 18:23.43 | bkw_ | brought to you by the nice guys that run Asterlink.com, Cluecon.com and more... |
| 18:23.53 | harryvv | :) |
| 18:24.06 | bkw_ | CLUECON registration is open btw.. |
| 18:24.17 | bkw_ | http://www.cluecon.com |
| 18:24.18 | harryvv | why the wording of clue? |
| 18:25.00 | harryvv | At first thought it was out of sarcasim :) |
| 18:25.02 | bjohnson | Colonel Mustard did it in the library with a candlestick |
| 18:25.08 | bjohnson | (the sick bastid) |
| 18:26.22 | bkw_ | harryvv, why? |
| 18:26.50 | darwin35 | bkw in the show with a bar of soup |
| 18:26.55 | darwin35 | soap |
| 18:27.00 | darwin35 | lol |
| 18:28.01 | harryvv | bkw, well there is alot of sarcasim and thought at first before going to that domain it was for people that did not have a clue :) |
| 18:32.23 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 18:32.47 | buddah | is there anyone that might be able to help me with a Caller ID issue? |
| 18:33.27 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) |
| 18:34.07 | Flyboy6440 | duh.. that was too easy... |
| 18:34.16 | Flyboy6440 | i should have figured that out sooner.. |
| 18:35.17 | Flyboy6440 | thanks shido6 was able to implement it by using a ## transfer option :) |
| 18:36.19 | blitzrage | Polycom IP500 question, in the Registration menu (web interface) under the Identification section, what is the Address field for? |
| 18:36.40 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm |
| 18:36.53 | ManxPower | blitzrage: want my phone specific configs? |
| 18:37.11 | blitzrage | ManxPower: I just want to document what the field is for, as it has confused me :) |
| 18:37.29 | blitzrage | I thought it said Username or something at first, but I've just noticed thats not what it says, but the phoen works fine :) |
| 18:38.09 | ManxPower | blitzrage: no idea. I just the text config files with the min number of settings needed. |
| 18:38.24 | blitzrage | ahhhh, yah, I need to document the web interface |
| 18:38.25 | ManxPower | http://pastebin.ca/11679 |
| 18:38.30 | CyberKnet | bkw_: how goes the day? |
| 18:38.35 | blitzrage | thanks though |
| 18:38.43 | blitzrage | anyone else who uses Polycoms? |
| 18:38.45 | ManxPower | blitzrage: well if the phone can upload it's config via FTP then change the settings, look at the uploaded config file |
| 18:38.54 | shido6 | whats up blitzrage |
| 18:39.04 | blitzrage | see my question above shido6 :) |
| 18:39.26 | blitzrage | ManxPower: yah, but I'm documenting for someone else to provision the phones, so I have to document it :) |
| 18:39.51 | buddah | can anyone point me in the direction of an answer on how to set the outbound caller id for a polycom ip 500? |
| 18:40.16 | blitzrage | buddah: Registration > Display Name |
| 18:40.47 | buddah | well i have 3 ip 500s, all with the same display name, but they dont show the display name for caller ID |
| 18:40.57 | buddah | they show the reg.x.address it seems |
| 18:41.12 | durex | brb |
| 18:41.14 | durex | asterisks... |
| 18:41.23 | durex | having problem to specify a realm to a register in sip.conf |
| 18:41.39 | durex | realm and domain in [general] didn't work... |
| 18:41.45 | durex | any idea ? |
| 18:41.53 | ManxPower | blitzrage: I think that is either the button label or the callerid info |
| 18:42.25 | blitzrage | hrmmmmm, well the button label is Label... |
| 18:42.35 | blitzrage | and callerID info is Display Name afaik |
| 18:42.58 | blitzrage | I have no idea why there would be an address field in the Indentification seciton when all that stuff is lower in the Server 1 config |
| 18:44.47 | ManxPower | <PROTECTED> |
| 18:45.15 | ManxPower | reg.1.auth.password and reg.1.auth.userid are the SIP userid and password |
| 18:45.23 | blitzrage | I'm not doing it from the text file so I have no idea what you're talking about :) |
| 18:45.40 | buddah | yeah manx |
| 18:45.49 | ManxPower | blitzrage: I'm saying you can set the options in the web interface, then look at the generated text file. |
| 18:46.05 | blitzrage | which is stored where? |
| 18:46.11 | ManxPower | blitzrage: OR you could just read the Admin Guide. It has info on the web interface, phone interface, and text file config interface. |
| 18:46.23 | blitzrage | I looke din the admin guide and found nothing about the web interface |
| 18:46.36 | ManxPower | blitzrage: the phone will upload the text config file to it's boot server. Info in the Admin Guide |
| 18:46.53 | blitzrage | like I said, I'm not using any of that stuff :) |
| 18:47.15 | ManxPower | blitzrage: It does not have a seperate section of the manual. Each option in the first part of the manual is mentioned how to set via each of the three interfaces. |
| 18:48.00 | bkw_ | shido6, |
| 18:50.44 | darwin35 | BKW I am moving to Arkensaw |
| 18:51.06 | Qwell | arkensaw? |
| 18:51.20 | ManxPower | darwin35: Marrying your sister? |
| 18:51.23 | Juggie | bkw_, is there a way to control the source ports * uses for RTP? |
| 18:51.27 | darwin35 | hillbilly way of spelling |
| 18:51.28 | machinehd | Anyone using a maxtnt with asterisk? |
| 18:51.36 | ManxPower | Juggie: Did you look at rtp.conf? |
| 18:51.51 | Juggie | those are listen ports |
| 18:51.58 | ManxPower | Juggie: Are you sure? |
| 18:52.01 | Juggie | yes |
| 18:52.05 | Juggie | positive |
| 18:52.05 | bkw_ | Juggie, rtp.c |
| 18:52.05 | ManxPower | OK |
| 18:52.07 | bkw_ | er rtp.conf |
| 18:52.21 | ManxPower | Juggie: Well you and bkw_ can argue about it. |
| 18:52.27 | Juggie | bkw_, those are the ports * listens on for RTP to be returned |
| 18:52.34 | Juggie | i'm talking about when it sends packets to a client |
| 18:52.37 | Juggie | say xlite for example |
| 18:52.45 | Juggie | it sends to port 8000 with a source of like 3000ish |
| 18:52.54 | Juggie | not in the range in the rtp.conf |
| 18:53.04 | bkw_ | what client? |
| 18:53.11 | bkw_ | you need to control the source ports on the client too |
| 18:53.40 | Juggie | well, i'm fightting with nat |
| 18:53.50 | Juggie | when i can get the client to use the ports given in the SDP |
| 18:53.54 | bkw_ | well control the source ports on both ends |
| 18:53.56 | Juggie | it works because i have that range taken care of |
| 18:53.57 | bkw_ | and you have it made |
| 18:54.32 | coppice | Juggie: sounds like a storybook I was reading my son this week about a bully called Nathaniel :-) |
| 18:54.42 | Juggie | but, xlite likes to try and be helpful, so audio works for a split second sometimes, until it gets RTP from * with a souce port of 3000ish |
| 18:54.48 | Juggie | then xlite starts sending to that port instead |
| 18:54.54 | Juggie | firefly does the same thing |
| 18:55.26 | Juggie | when clients follow the SDP it works. |
| 18:55.31 | Juggie | its when they dont thats bad. |
| 18:55.37 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 18:55.53 | blitzrage | sounds like a client problem :) |
| 18:55.56 | Juggie | i think the firewall may be smart enough to keep the source ports open for return traffic, i have to go play with it some more |
| 18:56.07 | Juggie | blitzrage, thats true |
| 18:56.09 | blitzrage | I'm not sure how you can control the source ports of the remote end if it doesn't listen to SDP |
| 18:56.24 | coppice | Juggie: if you can understand what teh SDP specs mean. I find them pretty woolly in places. |
| 18:56.35 | Juggie | it all depends on what * chooses to use as its source port for the RTP to the client |
| 18:56.37 | blitzrage | coppice: really? seems pretty straight forward to me... |
| 18:56.44 | Juggie | whatever * uses, thats what it will do. |
| 18:57.02 | Juggie | if the range was controlable, or it used source ports within the rtp range |
| 18:57.05 | Juggie | life would be good. |
| 18:57.26 | coppice | blitzrage: I am pretty unclear exactly what the T.38 stuff should really look like, and I don't seem to be alone from the implementations I have seen |
| 18:57.51 | bkw_ | thats what I keep trying to tell people |
| 18:57.51 | bkw_ | hehe |
| 18:58.03 | bkw_ | t.38 == HELL |
| 18:58.17 | blitzrage | coppice: oh yah, I avoid that stuff like the plague :) |
| 18:58.36 | coppice | yeah. I think everything about T.38 was developed by sheep farmers |
| 18:58.41 | blitzrage | lol |
| 18:58.44 | bkw_ | haha |
| 18:58.52 | Juggie | the firewall should be smart enough anyways to keep the source port open for return for RTP, its jst needs to be configed proper... but its isa2004 what do you expect |
| 18:59.10 | blitzrage | I'd like to go over the T.38 specs and just use a whiteboard and create a conversation and write out what the packets *should* look like |
| 18:59.39 | bkw_ | I think they leave a lot of the stuff up to the implementor |
| 18:59.40 | bkw_ | right? |
| 19:00.01 | *** join/#asterisk alvis (Alvis@200.105.128.59) |
| 19:00.08 | coppice | Yeah. like everything but the ASN.1 |
| 19:02.09 | blitzrage | thats stupid |
| 19:02.15 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 19:02.27 | blitzrage | what kind of a "standard" leaves it up to the implementor... if thats the case, whats the point of the standard |
| 19:02.41 | blitzrage | other than to make the ITU rich :) |
| 19:04.03 | coppice | Well, standards traditionally only specify the absolute minimum needed for interoperability, and try to obfuscate everything else. T.38 doesn't even do that |
| 19:04.26 | ManxPower | coppice: write your own spec, callit T.39 and publish it. |
| 19:04.36 | CyberKnet | coppice: heh. Not Fun(tm) |
| 19:04.37 | ManxPower | or T.38++ with everything specified. 8-) |
| 19:05.32 | bkw_ | well lets do it the asterisk way :P |
| 19:05.36 | bkw_ | wait nm |
| 19:05.37 | bkw_ | lets not |
| 19:06.02 | jets | hahaha break it and argue about eerything ? :) |
| 19:06.23 | ManxPower | write our own and not document it? |
| 19:06.25 | coppice | nah. write and implementation, and claim its the spec |
| 19:06.54 | jets | speaking of no documentation where does a queue timeout go to, the enxt priority? |
| 19:06.56 | tzanger | nobody here knows a little analog electronics, do they? |
| 19:07.07 | tzanger | the gain-bandwidth product of a bipolar transistor... is it linear? |
| 19:07.14 | blitzrage | .... yes |
| 19:07.23 | blitzrage | lol |
| 19:07.24 | coppice | tzanger: yes. a lot. advice is $300 per hour |
| 19:07.26 | *** part/#asterisk Flyboy6440 (~Bobo@192.76.82.90) |
| 19:07.26 | tzanger | i.e. if the GBP is 300MHz, then at DC it's gain is 300, and at 300MHz its gain is 1 |
| 19:07.55 | CyberKnet | hahaha |
| 19:07.55 | blitzrage | pay up biatch! :) |
| 19:07.57 | tzanger | coppice: ha, that important it isn't, I'm just trying to remember... but AFAIK the GBP isn't linear |
| 19:08.13 | blitzrage | with a name like Tony, watch your knees |
| 19:08.28 | tzanger | oh and FYI, nortel's optical links appear to be E1s that run at 2.56MHz |
| 19:08.49 | *** join/#asterisk L|NUX (~linux@202.5.145.54) |
| 19:08.58 | coppice | A transistor doesn't truly have a GBP. An amp built from the transistor does, and the GBP of that is linear |
| 19:09.47 | tzanger | ?? a bipolar transistor operated in its linear region is an amplifier, is it not? |
| 19:10.35 | tzanger | I was certain I could have driven a PN2222A at 2.56Mhz without trouble but it appears not. I wonder if a TIP120 can handle that frequency :-) |
| 19:10.43 | coppice | yep, but now we are constrained to linear behaviour, and GBP is a straight line |
| 19:10.46 | tzanger | failing that I'll just have to break down and get a small mosfet |
| 19:10.54 | tzanger | coppice: ahh okay |
| 19:11.18 | tzanger | so if I'm operating it as a switch the Max. switching frequency is nowhere near the GBP |
| 19:11.22 | tzanger | as I'm saturating it |
| 19:11.34 | tzanger | it's been quite some time since I've been playing with this stuff, as you can no doubt tell |
| 19:11.45 | tzanger | bjohnson: I'm the student, coppice's score would be off the chart |
| 19:11.49 | blitzrage | I haven't done any of that since highschool |
| 19:12.09 | coppice | "you can't have infinite GBP" is the electronics equivalent of mechanical engineers saying "you can't have perpetual motion" (although with sufficient curry it can seem like you do) |
| 19:12.13 | blitzrage | and we can all imagine how much I'd learned in highschool :) |
| 19:12.17 | tzanger | blitzrage: :-) I love analog design, I'm just a little rusty |
| 19:12.37 | blitzrage | tzanger: me too, I used to build guitar pedals in highschool :) |
| 19:12.42 | CyberKnet | I have suffered stunted perpetual motion from curry before =) |
| 19:12.49 | blitzrage | curry is the best |
| 19:13.01 | tzanger | I'm not asking for infinite GBP, just a transistor I have in the parts bin with sufficient switching frequency when driven into saturation :-) |
| 19:13.04 | CyberKnet | curry is the absolute best way to flush your system. |
| 19:13.19 | CyberKnet | providing of course that you have a REALLY stout toilet bowl. |
| 19:13.20 | tzanger | I thought I was perhaps just not giving it enough base current but I now know that's not hte case |
| 19:13.27 | tzanger | I don't do curry |
| 19:13.31 | coppice | I spent a few hours in an Indian hospital the other week after some fish curry :-( |
| 19:13.46 | blitzrage | ewwww |
| 19:13.51 | CyberKnet | coppice: aaah =( Now that does not sound like fun. |
| 19:13.53 | blitzrage | not a fish fan... |
| 19:13.58 | CyberKnet | I've never had a fish curry. |
| 19:14.07 | tzanger | a pipe length tester for TDM400P... at first I thought you turned it into a TDR :-) |
| 19:14.10 | CyberKnet | doesn't sound like I ever will, either. |
| 19:14.19 | CyberKnet | I nice green curry would be good though. |
| 19:14.55 | *** join/#asterisk TonyM (~softins@adsl-solo-80-168-226-67.claranet.co.uk) |
| 19:15.04 | coppice | a really good thai green curry - through away the meat, and just drink the gravy :-) |
| 19:15.09 | tzanger | oh, coppice: apparently zaptel from 1.0.7 does NOT CPU spike and works quite nicely with tx/rxfax |
| 19:15.14 | CyberKnet | tzanger: I heard some people giving negative accounts of the TDM400P when operated in the 1xFXO 1xFXS combination. Is there any merit to that? |
| 19:15.20 | tzanger | I have not confirmed it myself yet |
| 19:15.29 | tzanger | CyberKnet: I hope not, I've sold 3 of them :-) |
| 19:15.57 | CyberKnet | I want one, but can't justify the price when I can get a SPA-3000 for half the cost. |
| 19:16.11 | TonyM | Is JerJer about? |
| 19:16.18 | tzanger | JerJer: you about? |
| 19:16.22 | tzanger | (was it that hard??) |
| 19:16.32 | CyberKnet | tzanger: heh |
| 19:17.10 | tzanger | I've stolen two optical transcievers from the norstar system, and looped them to a pair of MAX490s (full-duplex RS485 transcievers) |
| 19:17.35 | tzanger | I've got it looping back and the singal looks good but I seem to be baffling these PN2222A transistors with a 2.5MHz clock |
| 19:17.51 | tzanger | 680 ohm resistor from the output of a HC14 to the base of the transistor |
| 19:18.13 | tzanger | one side of the ressitor, 2.5MHz squarewave, other side, about 0.7v |
| 19:18.19 | tzanger | the transistor's permanently saturated :-) |
| 19:18.54 | coppice | if the HC14 is switching it shouldn't be |
| 19:18.57 | tzanger | so I stuck a 25ohm resistor across the 680 ohm to really bop the base around... it worked to an extent but I was obviously doing it a disservice :-) |
| 19:19.16 | tzanger | well 4.3V across 680 ohms is 6mA base current |
| 19:19.35 | tzanger | I'm guessing the gain of the transistor is just too small at 2.5Mhz to really move the C-E current |
| 19:19.58 | tzanger | so I could either use a darlington (but they're slower) or break down and use a FET |
| 19:20.04 | coppice | its not the gain. its the resistor and the charge storage in saturation |
| 19:20.37 | tzanger | I'm not getting rid of charge in the B-E capacitance fast enough? |
| 19:21.32 | tzanger | the E of the transistor's just to common, the collector goes to a 50 ohm resistor to +5 and also to the A of the fiber optic emitter (whose K goes to common) |
| 19:21.49 | tzanger | so there's a fair whack of current it has to shunt when turning on |
| 19:22.24 | shido6 | brb |
| 19:22.27 | shido6 | need bay leaves |
| 19:22.41 | shido6 | u still lookin for pb&j info tzanger ? |
| 19:22.56 | tzanger | which is also why I can't just take the HC14 output (I have three paralleled up to make sure I have lots of drive) to the LED... the HC14 can only source about 25mA per inverter, but no real stats on what the entire chip can |
| 19:23.22 | tzanger | shido6: no just zaptel stuff, bkw seems to think that __do_dtmf blocks when sending DTMF |
| 19:23.33 | tzanger | I personally can't see it but at this point I haven't got a better idea |
| 19:26.58 | *** join/#asterisk mflorell (~mattf@rrcs-24-173-158-34.se.biz.rr.com) |
| 19:28.43 | *** join/#asterisk lattice (~lattice@S010600045ad57bb6.vc.shawcable.net) |
| 19:28.51 | tzanger | coppice: I did an opto design a while back that kept the phototransistor collector current change as small as possible to speed it up |
| 19:28.59 | tzanger | I was hoping I woudlnt' have to do something similar here |
| 19:29.56 | CyberKnet | heh. Verilan are selling a "TDM11B" bundle that has "One (1) TDM400P, One (1) FXS Module (green), Three (1) FXO Modules (red)" |
| 19:30.10 | CyberKnet | I'm guessing that you dont get three FXO =) |
| 19:30.44 | tzanger | CyberKnet: that's just a dev kit lite |
| 19:31.34 | CyberKnet | tzanger: Yeah, I saw the same bundle on digiums website and the contents. Just amusing that they say Three (1) FXO. |
| 19:31.39 | tzanger | haha |
| 19:31.45 | shido6 | heh |
| 19:31.46 | tzanger | Three (1) I got it now |
| 19:31.56 | CyberKnet | tzanger: aaaah... you didn't notice =) |
| 19:32.05 | tzanger | I look at TDM11B and know what it is |
| 19:32.13 | CyberKnet | Yeah, I know what it is too. |
| 19:32.21 | CyberKnet | digium will only sell it to developers. |
| 19:32.28 | CyberKnet | I'm sure they have their reasons. |
| 19:32.51 | tzanger | CyberKnet: ?? |
| 19:32.54 | tzanger | you're on crack |
| 19:32.54 | CyberKnet | Verilan + shipping is the same cost as digium - shipping though. |
| 19:32.59 | CyberKnet | tzanger: I assure you I am not. |
| 19:33.06 | tzanger | CyberKnet: the TDM11B can be purchased by anyone |
| 19:33.22 | CyberKnet | Not according to digium's online store. |
| 19:33.36 | CyberKnet | "Notes: This package is exclusively for Asterisk Developers" |
| 19:33.45 | *** part/#asterisk alvis (Alvis@200.105.128.59) |
| 19:33.46 | CyberKnet | http://store.yahoo.com/asteriskpbx/newitastdmde.html |
| 19:34.04 | CyberKnet | Now, they may ship it regardless... |
| 19:34.16 | CyberKnet | but the page says developers only. |
| 19:34.35 | tzanger | http://www.digium.com/index.php?menu=devkit-fxofxs |
| 19:34.36 | tzanger | where |
| 19:34.44 | CyberKnet | click "Order Online" |
| 19:34.44 | tzanger | "Digium now offers an alternative to the Asterisk Developer's Kit Lite and the Developer's Kit TDM to Asterisk developers and for people who wish to integrate Asterisk as a SOHO (Small Office / Home Office) PBX." |
| 19:35.04 | tzanger | "Notes: This package is exclusively for Asterisk Developers. " |
| 19:35.05 | tzanger | ahh |
| 19:35.22 | CyberKnet | nb: you can withdraw your crack accusation now =) |
| 19:35.37 | tzanger | CyberKnet: yes I do withdraw that... the website's on crack |
| 19:35.47 | CyberKnet | tzanger: that may be the case. I couldn't say. |
| 19:35.53 | marky | if i debug when i dial a phone number....and it adds 400 to the end? |
| 19:37.00 | marky | 400 is the prefix of my outbound trunk..not sure if that makes a difference |
| 19:38.06 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 19:38.30 | bjohnson | might |
| 19:38.39 | bjohnson | your dialplan would be where to fix it |
| 19:38.53 | CyberKnet | sup slePP |
| 19:39.11 | slePP | my internet sucks ass |
| 19:39.43 | CyberKnet | slePP: remove ass from internet. internet will no longer be able to suck it. |
| 19:40.17 | bjohnson | slePP: clean up your internet. There's pron all over it |
| 19:40.40 | slePP | who do you think put the porn there? :P |
| 19:41.08 | CyberKnet | slePP: sorry 'bout that porn, buddy =) |
| 19:41.20 | file | slePP porn! |
| 19:42.26 | slePP | slepp porn rules |
| 19:42.55 | CyberKnet | slePP: porn owned by slePP, or porn of slePP? |
| 19:43.14 | slePP | both |
| 19:43.41 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 19:43.44 | CyberKnet | gah. I'll pass. |
| 19:44.42 | CyberKnet | I have no desire to see slePP handling his Tiny286 |
| 19:44.45 | CyberKnet | no sir. |
| 19:44.47 | CyberKnet | ;) |
| 19:45.17 | file | awwww |
| 19:48.36 | *** part/#asterisk Cresl1n (~matt@216.207.245.23) |
| 19:50.16 | file | slePP is uber-sexy |
| 19:50.26 | CyberKnet | heh |
| 19:50.53 | file | mount /dev/slepp |
| 19:51.51 | CyberKnet | dd if=/dev/file of=/dev/slep |
| 19:52.05 | CyberKnet | erm... append one more p for me. |
| 19:52.08 | file | haha |
| 19:52.25 | *** join/#asterisk wasim (~wasim@203.81.201.78) |
| 19:53.09 | slePP | mmm... coke |
| 19:53.53 | file | bad slePP bad |
| 19:54.37 | coppice | tzanger: I updated to the latest CVS for zaptel. It seems my new test program doesn't see such wild jumps in he path length, but I still see some |
| 19:54.49 | tzanger | hmm |
| 19:54.57 | tzanger | vmstat doens't show CPU spiking every 5s or so? |
| 19:56.19 | coppice | can't see any spikes now, but I found that behaviour rather erratic anyway |
| 19:56.33 | tzanger | hmm interesting |
| 19:57.29 | mflorell | I've been trying to add a new Manager action to allow Bridging of 2 existing channels |
| 19:57.40 | mflorell | I've hit a wall, any developers out there? |
| 19:58.03 | coppice | what? behind the wall? |
| 19:58.11 | tzanger | the "all-ones" AIS pattern should look like a 50% duty cycle square wave of 2.048MHz for E1 and 1.544MHz for T1, shouldn't it? |
| 19:58.35 | mflorell | here's the code: http://astguiclient.sourceforge.net/ADDTO_res_features.c |
| 19:58.35 | TonyM | mflorell: not a real developer, but I try to hack - what's the issue? |
| 19:58.37 | coppice | no |
| 19:58.58 | tzanger | 193 1s sent 8000 times a second, or 257 bits 8000 times a second? |
| 19:59.01 | tzanger | (T1/E1) |
| 19:59.07 | mflorell | TonyM: me too, but this has been bugging me for a while so I decided to give it a shot |
| 19:59.32 | coppice | tzanger an E1 has a three level code |
| 19:59.58 | tzanger | hmm I need to read up on E1 |
| 20:00.08 | mflorell | I've got the two channels MASQd and then brodged just fine to the point they are talking to each other and the other 2 channels are dropped |
| 20:00.10 | TonyM | mflorell: I was going to have to do it in the near future, so I'm interested. Just pulled your code up on the screen. |
| 20:00.24 | *** join/#asterisk ClayReiche123 (fwuser@mail.accxx.com) |
| 20:00.28 | coppice | hence the name HDB3 |
| 20:00.43 | mflorell | then it kind of freezes |
| 20:01.30 | mflorell | I really just need to understand what is keeping ast_channel_bridge from returning |
| 20:01.37 | tzanger | coppice: hmm |
| 20:01.47 | ClayReiche123 | I'm noticing poor audio quality in conjunction with this NOTICE. I have already disabled silence suppression on the endpoint I'm in control of and it didn't seem to help... frame.c:128 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 20:01.52 | mflorell | and why the 2 dropped channels sometimes get stuck in channel_walk_locked mode |
| 20:02.08 | tzanger | HDB3 is just a variant of AMI is it not? |
| 20:02.25 | *** join/#asterisk Xen^ (~linux@202.5.145.54) |
| 20:02.31 | tzanger | it doesn't seem to be any different in terms of 1s and 0s, just inversions when transmitting on copper |
| 20:02.31 | ClayReiche123 | Can anyone tell me what I can do to fix it? |
| 20:02.42 | tzanger | (there is no -light and +light on fiber, obviously) |
| 20:02.48 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 20:03.14 | TonyM | mflorell: ah, ok. Sounds like you already know more about it than I do! What other part of Asterisk did you take ideas from? I was going to look at the code around the manager "Link" event. |
| 20:03.34 | syle | what is a good ip phone? |
| 20:04.04 | Zeeek | at what price? |
| 20:04.05 | mflorell | TonyM: you actually can't use anything currently in the manager, you cannot even code it into manager.c without a lot of other junk |
| 20:04.12 | *** part/#asterisk muntz (~msh@acheron.hsd1.ma.comcast.net) |
| 20:04.18 | syle | well no more than 300 would be nice |
| 20:04.18 | TonyM | mflorell: I just need to go out for 15-20 min to take a visitor home... will you still be around? |
| 20:04.28 | Zeeek | polycom |
| 20:04.42 | coppice | tzanger: fibre uses pulse operation, so is always a binary code. most wire codes are 3 level. It helps with whitening, and to reduce DC |
| 20:04.51 | mflorell | TonyM: I grabbed some stuff from res_features, app_dial.c, app_queue.c and channels.c |
| 20:04.59 | syle | i was thinking the cisco 7960 from the asterisk book |
| 20:05.11 | *** join/#asterisk Xen^ (~linux@202.5.145.54) |
| 20:05.15 | mflorell | TonyM: and res_features actually seems like a logical place for this to reside |
| 20:05.30 | tzanger | coppice: right I understand but that doesn't answer my AIS signal question :-) it's described as "unframed all-ones" which to me should look like a 50% duty squarewave of the clock frequency |
| 20:06.01 | tzanger | even on copper though, it's all-ones so AMI, B8ZS or HDB3 would look like +-+-+-+-+-- |
| 20:06.01 | coppice | tzanger: I just go a big jump in the loop length with the latest zaptel |
| 20:06.03 | coppice | 508 -> 516 -> 660 samples |
| 20:06.21 | mflorell | TonyM: this is about 3 half-days of playing around with this. I've tried lots of different ways of doing this. I had one simple way working, but it only worked between 2 IAX channels and only if they were connected to Playback |
| 20:06.24 | tzanger | coppice: if you've already got this up and running, would you be able to pull 1.0.7 and see what it does? |
| 20:06.41 | RoyK | sipgw1:~ # ps axvf|head -1 && ps axvf|grep -v grep|grep asterisk\ \-v|head -1 |
| 20:06.42 | RoyK | <PROTECTED> |
| 20:06.42 | RoyK | 13766 ? S< 0:15 0 605 1392038 1272608 61.3 \_ asterisk -vvvg -c |
| 20:06.44 | coppice | tzanger when coded in HDB3 is doesn't |
| 20:06.44 | RoyK | tamtitam...... |
| 20:06.56 | tzanger | coppice: ?? I'm looking here http://www.erg.abdn.ac.uk/users/gorry/course/phy-pages/hdb3.html |
| 20:07.03 | coppice | tzanger: if 1.0.7 is OK, shouldn't the latest CVS b OK? |
| 20:07.08 | mflorell | TonyM: I'll be here |
| 20:07.11 | ClayReiche123 | ...any takers...? |
| 20:07.14 | tzanger | HDB3 only fucks with how zeroes are sent (specifically, 4 zeroes) |
| 20:07.22 | *** join/#asterisk xlyz (~xl@213-140-17-96.fastres.net) |
| 20:07.28 | tzanger | 1s are always sent alternate-mark, at least if I'm reading this correctly |
| 20:07.38 | *** part/#asterisk xlyz (~xl@213-140-17-96.fastres.net) |
| 20:08.21 | tzanger | coppice: also, 1.0.7 is not CVS HEAD :-) someone here ysterday said their debian package for zaptel was receiving faxes all day for weeks with no problems |
| 20:08.28 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 20:08.33 | coppice | tzanger: before the HDB3 coder, yes. not on the wire |
| 20:08.38 | tzanger | the debian package is just 1.0.7, but there is a patch I haven't looked at yet |
| 20:09.26 | syle | how about iax phones? |
| 20:09.46 | coppice | has 1.0.7 been out for weeks? :-\ |
| 20:09.56 | darwin35 | months |
| 20:09.58 | Zeeek | syle I have a few of those |
| 20:09.59 | *** join/#asterisk AsteriskNoob (AsteriskNo@207-114-232-10.gen.twtelecom.net) |
| 20:10.04 | AsteriskNoob | afternoon all |
| 20:10.13 | ClayReiche123 | Hi |
| 20:10.47 | tzanger | coppice: forgive my ignorance, but could you explain? As I said, my understanding (which I'm willing to accept is wrong) is that HDB3 sends 1111 as + - + -... and 1s are always sent as the alternate mark of the last 1. zeroes are sent as the lack of mark, unless 4 are encountered, in which case a bipolar violation is sent as the 4th zero, and when eight zeores are sent the you have the first zero of teh quad turning into a 'balancing" BPV... so a singal |
| 20:10.48 | *** join/#asterisk Config_t (~jay@68-109-244-70.perimetercenter.net) |
| 20:11.40 | AsteriskNoob | so, i got an interesting issue |
| 20:11.41 | *** join/#asterisk L|NUX (~linux@202.5.145.54) |
| 20:11.49 | AsteriskNoob | i switched from 3 X100P's to a PRI circuit |
| 20:11.54 | Zeeek | http://www.iareaphone.com/ShoppingCart/catalogue_enterfromstatic.asp?ProductSet=10273 |
| 20:11.59 | Zeeek | syle ^^^ |
| 20:12.01 | *** join/#asterisk PBXtech (~nik@67.107.241.9.ptr.us.xo.net) |
| 20:12.02 | *** join/#asterisk docelm0 (~docelm0@67.106.194.90.ptr.us.xo.net) |
| 20:12.34 | AsteriskNoob | and i used to be able to use the dial string of 1-XXX-XXX-XXXX on my phones, and the - meant nothing to a DTMF based system so it worked, with the PRI however any calls with a - in them is rejected.... How can I remove all the -'s before dialing? |
| 20:12.37 | docelm0 | Does anyone know of a SIP<->H323 converter or a module that actually works with asterisk 1.0.7? |
| 20:13.07 | coppice | tzanger: Here's a pretty picture http://www.trendcomms.com/multimedia/training/broadband%20networks/web/main/m2/temari/seccio8/hdb3.htm |
| 20:13.23 | syle | isn;t there a flashing light in case you have messages in asterisk on that phone? |
| 20:13.23 | Zeeek | AsteriskNoob there are string handling apps - read the doc (I don't remember the name) |
| 20:13.26 | syle | looks pretty plain |
| 20:13.33 | Zeeek | syle - no MWI |
| 20:13.41 | syle | damn, out of question then |
| 20:13.50 | *** join/#asterisk focks (~craig@nsc66.147.95-93.newsouth.net) |
| 20:13.58 | Zeeek | that' about the best iaxphone I've seen so far |
| 20:14.08 | syle | without MWI i might as well just have an analog phone |
| 20:14.16 | docelm0 | Does anyone know of a SIP<->H323 converter or a module that actually works with asterisk 1.0.7? |
| 20:14.31 | focks | what would I need to add to this to wait 1 second before dialing so i don't have problems with the telco not getting all the digits? exten => _NXXXXXX,1,Macro(dialout-trunk,1,${EXTEN}) |
| 20:15.00 | tzanger | coppice: that picture just verifies what I'm saying |
| 20:15.07 | tzanger | if I sent nothing but ones, I get a squarewave |
| 20:16.36 | ClayReiche123 | Can I get some help? I want to know if I can disable whatever it is that drops g729 packets due to VAD... |
| 20:16.44 | ClayReiche123 | frame.c |
| 20:17.08 | ClayReiche123 | frame.c:128 ast_smoother_feed |
| 20:17.20 | coppice | tzanger: oh, yeah, sorry. I'm so used to seeing three levels I forgot that continuous ones actually works out to a square wave. |
| 20:17.28 | tzanger | ok... whew :-) |
| 20:18.01 | tzanger | so an AIS signal on an E1 looks like a 2.048MHz squarewave |
| 20:18.14 | tzanger | I'm seeing a 2.56Mhz squarewave on this Norstar system |
| 20:18.28 | coppice | a 1.024MHz square wave |
| 20:19.07 | syle | is it possible to setup an extension to access your voicemail box without having to put in a username password? |
| 20:19.14 | blitzrage | syle: yep |
| 20:19.15 | tzanger | hmm |
| 20:19.18 | Meaty | I have a channel bank configured on asterisk dans i want to user a special (second) ring for my analogue phone. Is it possible with asterisk ? |
| 20:19.27 | tzanger | your'e right |
| 20:19.32 | syle | please do tell, cause its a real pain in the ass lol |
| 20:19.37 | tzanger | so wtf is this norstar system sending then, heh |
| 20:19.37 | blitzrage | syle: show application voicemail |
| 20:20.06 | blitzrage | syle: your hint - '8500' => 1. VoicemailMain(${CALLERIDNUM}) |
| 20:20.34 | blitzrage | syle: the rest can be obtained via show application voicemail |
| 20:20.38 | coppice | one of the schemes for junctiopn lasers ends up with something like 2.56MHz for an E1. Its years since I touched it, though (used to have a team designing them once upon a time). |
| 20:21.21 | syle | blitzrage: VoicemailMain(s${CALLERIDNUM}) is what i have but i don;t think that is working cause of the analog phone possibly |
| 20:21.47 | coppice | 2.56 = 1.25*2.048. Its something about stuffing 5 things were there were 4 |
| 20:22.01 | ClayReiche123 | Can I turn VAD off on asterisk? |
| 20:22.18 | *** join/#asterisk dos000 (~dos000@66.11.173.123) |
| 20:22.27 | blitzrage | ClayReiche123: you turn it off on the phone |
| 20:22.36 | tzanger | coppice: oh dear |
| 20:22.40 | syle | i know in incomming i can do : exten => s/phonenumber,1,Wait(1) |
| 20:22.42 | dos000 | anyone knows the cost of the proprietary ss7 stack from digium ? |
| 20:22.49 | tzanger | it appears my plans to take over nortel are a little overambitious :-) |
| 20:22.53 | syle | etc |
| 20:23.00 | tzanger | I was hoping to convert the optical signal to copper and feed it into a Te405 |
| 20:23.21 | dos000 | tzafrir, nortel .. pfft ... it takes courage to work there these days ! |
| 20:23.57 | coppice | nortel was a really great place when I worked there. sad to see what has happened. |
| 20:23.57 | tzanger | built the circuit to do that and just fed it back over to the other optical interface... but as I said I was seeing 2.56MHz squarewave on what I'd call the AIS so it was throwing me |
| 20:24.15 | ClayReiche123 | blitzrage: * appears to be doing its own. I have * in my media path and I'm getting this message. (along with poor audio) frame.c:128 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 20:24.45 | tzanger | coppice: do you have a name for this 2.56MHz E1 clock thing so I might do some more research? |
| 20:24.45 | coppice | You need optical to 3 level conversion. HDB3 is 3 level, whatever a continuous 1 might look like. |
| 20:24.57 | tzanger | yeah I ust found something |
| 20:25.01 | tzanger | DS21610 |
| 20:25.39 | blitzrage | ClayReiche123: I've never known * to have VAD built in... |
| 20:25.46 | tzanger | "converts T-carrier clock rates to E-carrier clock rates" and it mentioned 2.56Mhz too |
| 20:26.18 | tzanger | CLKIN of 2.56MHz, CLKOUT of 1.544 |
| 20:26.26 | tzanger | must read more :-) |
| 20:26.34 | tzanger | I just plugged in "2.56MHz E1" and that came out |
| 20:26.56 | *** join/#asterisk Juxt (~Juxt@64.135.20.202) |
| 20:27.16 | bjohnson | Meaty: yes |
| 20:27.20 | Juxt | good afternoon |
| 20:27.26 | Juxt | does anyone use firefly here? |
| 20:27.34 | bjohnson | Meaty: depends on phone hardware |
| 20:27.52 | coppice | It was 1988 the last time I worked with E1 over fibre :-\ |
| 20:28.03 | tzanger | aww come on that's not that long ago :-) |
| 20:28.04 | coppice | at nortel, too |
| 20:28.19 | Juxt | i can't seem to make delays when i add a number in contact |
| 20:28.20 | tzanger | I appreciate your help and insight so far though, it's really helped |
| 20:28.56 | Meaty | bjohnson: My phone is analogic basic phone |
| 20:29.19 | bjohnson | you want a different ring .. or two different rings? |
| 20:29.22 | syle | hate analog no MWI |
| 20:29.30 | bjohnson | syle: incorrect |
| 20:29.35 | Meaty | 2 different |
| 20:29.39 | Zeeek | some analog phones have mwi |
| 20:29.40 | *** join/#asterisk twilson (~terry@63.77.68.11) |
| 20:29.41 | Meaty | Is it not the channel that can this ? |
| 20:29.51 | Meaty | channel banks* |
| 20:29.51 | bjohnson | syle: I have 2 analog phones with MWI that worj with my SPA units |
| 20:30.27 | syle | lucky bastard, i have it but phone doesn;t seem to let me configure the MWI |
| 20:30.42 | coppice | syle: many analog phones have MWI |
| 20:31.11 | Romik | any advice on it? firefly with asterisk 1.07 | May 13 16:34:15 NOTICE[19682]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/livevoip/3 of format speex since our native format has changed to ulaw |
| 20:31.11 | syle | not the panasonic kx-tg5110 i have |
| 20:31.11 | Romik | <PROTECTED> |
| 20:31.15 | *** join/#asterisk drumkilla_ (~russell@user-24-214-77-225.knology.net) |
| 20:31.15 | *** mode/#asterisk [+o drumkilla_] by ChanServ |
| 20:31.23 | tzanger | anyway I gotta go get my kids, ttyl |
| 20:33.11 | ClayReiche123 | blitzrage: I'm only inferring from the NOTICE I'm getting... just looks like it's doing "something" to the media stream.... |
| 20:33.29 | blitzrage | drumkilla_: !!! |
| 20:33.42 | drumkilla_ | hey blitzrage !!!! |
| 20:33.51 | blitzrage | drumkilla_: how goes? |
| 20:33.55 | drumkilla_ | I have been at my apartment for 3.5 hours trying to get internet and cable hooked up |
| 20:33.56 | blitzrage | drumkilla_: pass everything? |
| 20:34.04 | drumkilla_ | ha, yeah, I passed |
| 20:34.11 | blitzrage | drumkilla_: haha, oh the joys of the first few days :) |
| 20:34.12 | blitzrage | nice |
| 20:34.14 | *** part/#asterisk makhtar (~ageller@mail3.bulletinnews.com) |
| 20:34.35 | *** join/#asterisk wizhippo (~Dude@CPE0080c816780d-CM0000390aca5e.cpe.net.cable.rogers.com) |
| 20:34.43 | drumkilla_ | blitzrage: what's up with you? |
| 20:34.43 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 20:34.58 | blitzrage | drumkilla_: trying to figure out how to setup auto-rollover redundency for Asterisk... :) |
| 20:35.18 | drumkilla_ | fun times |
| 20:35.30 | blitzrage | damnit, I was hoping you'd reply with, "oh thats easy, just use XXX" |
| 20:35.38 | drumkilla_ | well, ha |
| 20:35.39 | Meaty | bjohnson : I have foung |
| 20:35.41 | wizhippo | anyone have a click sound after a call is connected using tdm400's? |
| 20:35.50 | drumkilla_ | I mean, there are multiple ways |
| 20:36.03 | Meaty | bjohnson : Its 333,1,Dial(Zap/1r2) |
| 20:36.16 | Meaty | r2 mean ring 2 |
| 20:36.19 | drumkilla_ | blitzrage: tell me more about what you want |
| 20:36.23 | syle | wiz now you mention it i have heard that a few times |
| 20:36.24 | blitzrage | drumkilla_: basically, I want to have a list of servers, then if the call fails to go through on the first one ebcause the server is down, then try serv2, etc... |
| 20:36.24 | Meaty | Thanks for your help |
| 20:36.27 | syle | not very often though |
| 20:36.29 | Meaty | alls :P |
| 20:36.43 | drumkilla_ | blitzrage: well, in Asterisk, that's really simple, right? |
| 20:36.54 | blitzrage | drumkilla_: file says ${dialstatus} |
| 20:37.12 | blitzrage | I think I've buiilt it up too complex in my head, lol |
| 20:37.41 | drumkilla_ | mhm :p |
| 20:37.45 | wizhippo | I get it all the time. any idea how to get rid of it? it's sounds like another extensions is picked up and hung up. Almost like someone is listening in. |
| 20:37.47 | Juggie | anyone use iax with fwd? i can get into the thursday conference server, but i'm not getting registered to fwd... |
| 20:37.49 | drumkilla_ | that's simple dialplan stuff, man! |
| 20:38.07 | blitzrage | drumkilla_: whats a dialplan? |
| 20:38.34 | blitzrage | drumkilla_: but do the channels return a chanisunavail when its unavail, or congestion, thats the real question. I remember tzanger mentioning something about that previously :) |
| 20:39.14 | wizhippo | some one mentionioned it might be the tdm gards and how long they hold the voltages reversed until a call is established. |
| 20:39.36 | TonyM | mflorell: you still there? |
| 20:40.25 | kb1_kanobe | wizhippo: go you have the zaptel echotraining turned up? |
| 20:41.12 | wizhippo | i have it set to yes |
| 20:41.37 | drumkilla_ | blitzrage: well if the server is down, it would be unavail right? |
| 20:41.41 | drumkilla_ | blitzrage: no server to send congestion |
| 20:41.43 | blitzrage | well... it should be :) |
| 20:41.49 | drumkilla_ | indeed |
| 20:41.52 | kb1_kanobe | wizhippo ok, I just wondered if you'd set it to something else. no problem. |
| 20:41.53 | blitzrage | but I heard it sets congestion |
| 20:41.58 | blitzrage | I need to test to verify though |
| 20:42.01 | file | try and see |
| 20:42.05 | drumkilla_ | yah |
| 20:42.14 | wizhippo | when i compiled i turned on aggresive becasue I can here it echo when i type on my keyboard |
| 20:43.13 | kb1_kanobe | aggressive can do wierd things to the sound - did it noticbly improve your echo problem? |
| 20:44.34 | wizhippo | it got rid of the echo picked up from typeing |
| 20:44.54 | blitzrage | wizhippo: have you tried turning on echo training? (just a suggestion) |
| 20:45.00 | *** join/#asterisk jr99 (~jr99@adsl-065-005-202-014.sip.gnv.bellsouth.net) |
| 20:45.05 | wizhippo | ya it's on |
| 20:45.11 | jr99 | someone save me from broadvoice.. OMG they suck. |
| 20:49.38 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 20:50.01 | *** join/#asterisk r3d5un (~r3d5un@80.121.192.27) |
| 20:50.21 | pigpen | Hi all...How many digium 4 port fxo cards would I dare to put in a single P4 3.0Ghz Xeon 2 w/ 1GB ram? |
| 20:50.37 | Juxt | 4 |
| 20:50.37 | pigpen | about 20 sip clients |
| 20:50.44 | pigpen | that many?....really? |
| 20:50.55 | Juxt | yeah, then you'll run out of irqs |
| 20:51.06 | pigpen | k...the proc/mem can handle it? |
| 20:51.10 | Juxt | make sure it's a board where you can assign an irq to a pci slot |
| 20:51.14 | Romik | pigpen: put 2 x of quad T1 cards and connect 8 channels banks |
| 20:51.19 | Juxt | yeah not a prob at all |
| 20:51.26 | Juxt | in fact your box is an overkill |
| 20:51.31 | Juxt | by a factor of 10 or so |
| 20:51.39 | pigpen | cool... |
| 20:51.52 | pigpen | thanks...it has been hard to find a difiniatve answer.. |
| 20:52.22 | pigpen | Romik, that is what I was thinking about if I need more... |
| 20:52.31 | pigpen | who's channel bank do ya'll like? |
| 20:52.38 | bkw_ | ones that work |
| 20:52.47 | pigpen | true... |
| 20:52.54 | bkw_ | the kind that take t1's and turn them into FXO/FXS |
| 20:53.25 | *** part/#asterisk r3d5un (~r3d5un@80.121.192.27) |
| 20:53.37 | jr99 | how stupid would it be to take a vonage or packet8 adapter and put it into a FXO port on a digium card? |
| 20:53.37 | Romik | pigpen: i use zhones....cheap ones...i buy them for 169$ each |
| 20:53.52 | bkw_ | jr99, it about all you can do |
| 20:54.11 | mflorell | zhones are great, I even made a perl script to speed config them |
| 20:54.12 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 20:54.14 | pigpen | Romik, how many analog lines would it handle? |
| 20:54.16 | jr99 | sucks. packet8 and vonage are the only ones who have the DIDs I want.. |
| 20:54.27 | bkw_ | jr99, give me till the end of the day |
| 20:54.29 | jr99 | broadvoice does, but even with the right DIDs they don't work most of the time. |
| 20:54.31 | bkw_ | I might be able to help you |
| 20:54.32 | Romik | mflorell: could you share your script? |
| 20:54.33 | TonyM | Any chan_h323 gurus here? |
| 20:54.33 | pigpen | I would guess I would need to get a single pri card then.. |
| 20:54.36 | jr99 | heh |
| 20:54.46 | Romik | pigpen: each T1 is 24 lines |
| 20:54.50 | mflorell | sure, just a minute... |
| 20:55.20 | Romik | pigpen: they have 2 kinds... 1st = 8 FXO and 16 FXS or 2nd= 24FXS |
| 20:55.22 | mflorell | here's the zhone programmer script: http://astguiclient.sourceforge.net/experimental_code/Zhone_zplex_24s_program.pl |
| 20:55.25 | pigpen | sure...pri/chanalized T1 does 24 channels..but the zhones ...how many analog... |
| 20:55.29 | *** join/#asterisk NightHawke (~NightHawk@66.190.111.175) |
| 20:55.45 | *** part/#asterisk Fanguin (~Fanguin@p548F1EE9.dip0.t-ipconnect.de) |
| 20:55.57 | NightHawke | TDM cards, whats the pinouts for RJ11 pots |
| 20:56.08 | mflorell | the script is for the 24s zhone, not the one that is 8 and 16 |
| 20:56.17 | mflorell | although you could adapt it to your needs |
| 20:56.45 | mflorell | We had to program 8 of them, and after the first one I figured I'd save an hour if I programmed a script to do it |
| 20:56.50 | *** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 20:56.52 | pigpen | cool... |
| 20:56.53 | Romik | mflorell: what this script do? :) my system admin configure it via serial port.. |
| 20:56.54 | ClayReiche123 | drumkilla: do you know anything about this NOTICE? frame.c:128 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 20:57.04 | Tall-guy | anyone using asterisk with ComWave (voip provider)? |
| 20:57.10 | mflorell | you just need to config the network port via serial |
| 20:57.25 | Romik | mflorell: cool |
| 20:57.28 | BoRiS | Clay: Turn off VAD |
| 20:57.34 | mflorell | then it logs in and sets every analog port and T1 channel and crosses them together |
| 20:57.41 | BoRiS | VAD/Silence Suppression |
| 20:57.41 | ClayReiche123 | How? |
| 20:57.44 | Romik | mflorell: what rxgain and txgain do you use? |
| 20:57.52 | ClayReiche123 | On both endpoints? |
| 20:57.59 | mflorell | for Zhone channelbanks?, 0, 0 |
| 20:58.03 | BoRiS | On your sip phones |
| 20:58.12 | *** join/#asterisk bofh42 (~bofh42@pD9EC0849.dip0.t-ipconnect.de) |
| 20:58.21 | *** part/#asterisk Juxt (~Juxt@64.135.20.202) |
| 20:58.43 | NightHawke | TDM cards, whats the pinouts for RJ11 pots |
| 20:58.49 | Romik | mflorell: on zapata configuration? |
| 20:59.07 | ClayReiche123 | BoRiS: It's off on my Sip phones... I use Qwest as a Voip provider... they must have it on their gateway... |
| 20:59.38 | ClayReiche123 | BoRiS: support call to them I guess... I hope they will turn it off for me.... Thank you BoRiS! |
| 20:59.48 | BoRiS | np Clay :) |
| 21:00.03 | mflorell | Romik: signalling=fxo_ks |
| 21:00.43 | mflorell | Romik: and zaptel needs fxoks=1-24 |
| 21:01.00 | *** part/#asterisk ClayReiche123 (fwuser@mail.accxx.com) |
| 21:01.17 | mflorell | that is for 24 FXS Zhone though not the one with 8 FXO ports, that's a little different |
| 21:01.31 | niZon | BoRiS: what voip phones do you have? |
| 21:01.40 | Romik | mflorell: zapata.conf:signalling = fxs_ls |
| 21:01.41 | Romik | zapata.conf:signalling = pri_cpe |
| 21:01.41 | Romik | zapata.conf:signalling = fxo_ks |
| 21:02.16 | mflorell | I believe you can set it up to do loop-start, but never used it myself |
| 21:02.31 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 21:02.32 | mflorell | I always use Kewlstart for Channelbanks |
| 21:02.41 | mflorell | bwk_: got a minute to look at some code |
| 21:02.47 | devel | i just updated to the latest cvs (monday), and now dtmf (rfc2833) on my grandstream devices (bt and ht) doesn't work to asterisk (vm, transfer) but works extermally, and works fine for other devices (polycom, sipura, snom). is it just me? |
| 21:02.51 | Romik | mflorell: what your rxgain and txgain in zapata.conf? |
| 21:03.14 | mflorell | Romik: 0.0 and 0.0, but on one machine I have 1.0 and 1.0 |
| 21:03.27 | *** join/#asterisk my007ms (~mywaleed@84.36.14.135) |
| 21:03.36 | Romik | mflorrel: you do not hear echo? |
| 21:03.50 | mflorell | not with echo-cancelling turned on |
| 21:03.58 | mflorell | sometimes when it is off |
| 21:04.07 | mflorell | depending on the load on the machine |
| 21:04.15 | mflorell | so I just leave it on |
| 21:04.36 | BoRiS | hi niZon!!!! |
| 21:04.38 | BoRiS | file!!!!!!!!!! |
| 21:04.53 | BoRiS | slePPy!!!! |
| 21:05.00 | niZon | hey |
| 21:05.00 | niZon | lol |
| 21:05.03 | Romik | mlforell: we have echo....when call abroad..or voip...we have rxgain=0 and txgain=-6 |
| 21:05.17 | Romik | mflorell: i think we should put rxgain=-1 |
| 21:05.27 | Romik | what is this Kewlstart ? |
| 21:05.42 | mflorell | Romik: what trunks do you use?, do you have echo-cancelling turned on in zapata? |
| 21:05.45 | NightHawke | TDM cards, whats the pinouts for RJ11 pots |
| 21:06.07 | mflorell | Kewlstart is just another RBS protocol for signalling |
| 21:06.29 | mflorell | I think there is Groundstart, Loopstart, kewlstart |
| 21:06.30 | file | slePP: I see you! |
| 21:06.51 | file | actually I don't, but whatever! |
| 21:07.11 | Romik | mflorell: echocancel=yes |
| 21:07.11 | Romik | , echocancelwhenbridged=yes |
| 21:07.11 | Romik | , echotraining=yes |
| 21:07.11 | Romik | , echotraining=80 |
| 21:07.33 | mflorell | Romik: and you still get echo? |
| 21:07.55 | mflorell | Romik: What kind of trunks do you use? |
| 21:08.13 | Romik | mflorell: rare, but still, most calls routed via livevoip or voipjet |
| 21:08.40 | Romik | mflorell: even when i call from my ATA via my home asterisk to the office via speex ..i head very small echo. |
| 21:09.28 | mflorell | Romik: well, the echo would probably be on them then. We use T1s mostly and have very little echo, even the IAX2->GSM from remote offices usually don't have echo |
| 21:09.44 | Romik | mflorell: i hear my self...like i send very high signal...or something else... |
| 21:10.01 | Romik | mflorell: when we on PRI there no echo at all ...crystal clear |
| 21:10.05 | AgiNamu | slinear is exactly PCM is the same as in a normal wav file right |
| 21:10.40 | AgiNamu | Can socket() return 0 as a valid value? |
| 21:11.31 | Silik0n | damn T-Mo pisses me off |
| 21:11.34 | NightHawke | TDM cards, whats the pinouts for RJ11 pots |
| 21:12.23 | Silik0n | NightHawke: the middle pins are ring&tip... the actual pin numbers depend on if they used 4 6 or 8 wire jacks |
| 21:12.35 | Silik0n | (or 2 wire jacks) |
| 21:12.45 | NightHawke | will be wiring 2 pair/phine |
| 21:12.55 | NightHawke | phones are POTS types |
| 21:13.37 | NightHawke | and the jacks are RJ45 type |
| 21:14.39 | Romik | mflorell: where do you buy connection cable 25 pair to phone patch pannel? |
| 21:14.42 | NightHawke | so essentially the pairings are pretty much ITU standard |
| 21:15.12 | Silik0n | then you'll use pins 4&5 on the RJ45 for line 1 and pins 3&6 for line 2 if you have 2line pots phones |
| 21:15.33 | NightHawke | ty all! ^.^ |
| 21:15.48 | mflorell | Romik: we usually use a type 66 punch-down block with a standard 25-pair cable, you can get them cheap on ebay. then we punch down pairs to the phone jacks(RJ11) |
| 21:15.58 | Silik0n | romil: check your local cabling supply house they will have AMP 25pair paigtails |
| 21:15.59 | *** part/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net) |
| 21:16.30 | Romik | mflorell: what name of this cable? |
| 21:16.31 | mflorell | Romik: Graybar in the USA has them just about everywhere, but they aren't cheap |
| 21:16.34 | Silik0n | or you can get AMP to AMP cables and punchdown blocks prewired with the mating AMP connectors |
| 21:17.11 | mflorell | just a 25-pair female-female cable with Amphenol connectors on both ends |
| 21:17.14 | Silik0n | Romik: getting the from graybar or ${favorite_cable_supplier} is cheapre then paying a field tech to build them |
| 21:17.43 | Romik | mflorell: we use custom made 50 pin Amphenol connector to ethernet patch pannel.. it cost us $90 each... |
| 21:17.44 | Silik0n | but if this is for a home project... |
| 21:18.05 | mflorell | that's actually not bad |
| 21:18.14 | coppice | but getting people to spend hours hand assembling a couple they coulde buy for peanuts is so traditional :-) |
| 21:18.24 | mflorell | I've seen them for over $160 in some catalogs |
| 21:18.53 | Romik | coppice: i live not in states....for me order something in states...it simple pay it twice local.. |
| 21:19.33 | coppice | a 50 - 50 amphenol leads should be about $10-15 |
| 21:19.52 | *** join/#asterisk extremis (~extremis@cpe-24-175-55-177.houston.res.rr.com) |
| 21:19.53 | Silik0n | romik appears to be in Isreal |
| 21:20.09 | Silik0n | (which obviously i cant speel) |
| 21:20.27 | Romik | silik0n: hehehe |
| 21:20.31 | extremis | does anyone have a cdr-csv to postgress conversion script? The mysql import and conversion scripts don't handle my csv correctly so alot of the records are inserted with 0 for date and time |
| 21:21.36 | Romik | coppice: what about 66 punch-down block with a standard 25-pair cable? what name of this cable...? i just made search on ebay - not found anything with AMP 50 male connector |
| 21:23.42 | ManxPower | Romik: Amphenol |
| 21:24.05 | Corydon-w | And try graybar.com |
| 21:24.34 | *** join/#asterisk Vco (~Vco@S0106080020aa7650.wp.shawcable.net) |
| 21:24.41 | extremis | cdr-csv to postgres? anyone? |
| 21:25.07 | *** part/#asterisk mflorell (~mattf@rrcs-24-173-158-34.se.biz.rr.com) |
| 21:25.14 | BoRiS | Sorry extremis, I can't help you :( |
| 21:25.34 | bkw_ | extremis, its called PERL |
| 21:25.38 | bkw_ | use it and abuse it |
| 21:25.57 | BoRiS | :) |
| 21:26.01 | bkw_ | oh guess what |
| 21:26.04 | BoRiS | what? |
| 21:26.09 | Romik | greybar.com answer: No items matched "Amphenol". |
| 21:26.17 | bkw_ | asterisk has bugs |
| 21:26.25 | bkw_ | haha Romik for a moment I thought that was gaybar.com |
| 21:26.26 | bkw_ | haha |
| 21:26.38 | Romik | bkw: yes |
| 21:26.40 | outtolunc | siemon s66m |
| 21:27.17 | *** part/#asterisk wizhippo (~Dude@CPE0080c816780d-CM0000390aca5e.cpe.net.cable.rogers.com) |
| 21:27.33 | Romik | anybody can tell me product number for this "punch down blocks prewired with the mating AMP connectors"? |
| 21:27.36 | extremis | bkw: you have a perl script for csv to mysql by chance? |
| 21:29.08 | outtolunc | Cross Connect Hardware Siemon S66M4-24 Display item details for 284879 88233987 Field-Terminated M Series S66 Blocks |
| 21:29.43 | blitzrage | if someone calls into Asterisk via a PRI, dials digits, will Asterisk convert that inband to RFC2833 if the call is forwarded back out to another server, or will it stay inband? |
| 21:30.14 | bkw_ | extremis, its a few lines of code.. you can code it in a shakes of a lambs tail |
| 21:30.19 | RoyK | blitzrage: pri uses OOB, right? |
| 21:30.27 | bkw_ | correct |
| 21:30.37 | blitzrage | I suppose it should, thats right :) |
| 21:30.41 | extremis | bkw: nope |
| 21:31.12 | extremis | I'm not sure what invalid characters are |
| 21:31.14 | blitzrage | any ideas why I would be getting dupe DTMF being sent via RFC2833 then? |
| 21:31.25 | extremis | so I don't know how to properly sanitize it before trying to shove it in the db |
| 21:31.29 | Romik | outtolunc: but this one comes without 50AMP cable pre wired ? |
| 21:31.53 | extremis | but apparently the solutions that have been posted don't sanitize things correctly, but since I imported so many records its hard to track it down |
| 21:32.10 | extremis | I was hoping that someone else suffered that pain first |
| 21:32.33 | bkw_ | $line =~ s/\'/\\'/g; |
| 21:33.00 | outtolunc | romik, call graybar, tell the rep you want a siemon s66m punch down with 2 25pair male or female (you call) one CO on CPE side |
| 21:33.08 | *** join/#asterisk MasterYoda (~mnicholso@207.111.174.1) |
| 21:33.28 | extremis | bkw: what about " |
| 21:33.35 | MasterYoda | extconfig can be used for zaptel.conf correct? |
| 21:33.41 | extremis | other people are stripping it from the beginning and end and near ,'s |
| 21:33.56 | outtolunc | that of paypal me some $$ and i'll send you some <G> |
| 21:34.43 | Romik | this bug/problem i have since 1.05 (now 1.07) verision of asterisk - chan_zap.c:1472 zt_set_hook: zt hook failed: Device or resource busy for details: http://pastebin.ca/11698 |
| 21:34.54 | *** part/#asterisk MasterYoda (~mnicholso@207.111.174.1) |
| 21:35.17 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 21:35.24 | extremis | I'm also curious what ('s would do |
| 21:35.27 | extremis | and | |
| 21:36.37 | *** join/#asterisk ClayReiche123 (fwuser@mail.accxx.com) |
| 21:36.51 | ClayReiche123 | hi all |
| 21:40.33 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 21:40.35 | shido6 | one |
| 21:40.55 | ClayReiche123 | I have a deep/heady question... It appears that * is sending SDP information for g729a different than my upstream Voip provider is expecting. They are using a Sonus switch/gateway and they are expecting to see "a=fmtp:18 annexb=no" to determine g729a and * apparently sends "a=silenceSupp:off" to specify g729a. Is there any way for me to change how that is specified in the INVITE with SDP? |
| 21:41.06 | ClayReiche123 | ...maybe it's not so deep.... |
| 21:41.08 | ClayReiche123 | :) |
| 21:41.10 | extremis | other than INSERT,SELECT... what else should I grant the areski user to do? |
| 21:41.47 | file | ClayReiche123: a=silenceSupp:off is not for g729a, it's just saying that silence suppression is off |
| 21:42.12 | bkw_ | correct |
| 21:42.17 | channan | hi, anyone's using Broadvoice here? My broadvoice has been down most of the time in more than a week now. Many calls to Broadvoice (and emails) with no answer... What's going on??? |
| 21:42.22 | bkw_ | file do you have a util to convert gsm to g729 and g723.1 files? |
| 21:42.27 | bkw_ | I can't find the one I had |
| 21:42.56 | ClayReiche123 | file: Should * be using "a=fmtp:18 annexb=no" then? |
| 21:43.29 | *** part/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 21:44.11 | ClayReiche123 | file: I guess the question is, how do I get "a=fmtp:18 annexb=no" in my Invite with *? |
| 21:44.28 | *** join/#asterisk OzJames79 (~opera@CPE20320889-1842-1.gex.ncable.net.au) |
| 21:44.32 | *** join/#asterisk marky (emes@65.114.80.8) |
| 21:44.36 | marky | well maybe... |
| 21:44.53 | ClayReiche123 | ...or "can I"? |
| 21:45.23 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
| 21:52.01 | CyberKnet | bkw_: you have any ballpark on when you'll get DID LNP in Oklahoma? |
| 21:54.58 | bkw_ | CyberKnet, don't know yet |
| 21:55.02 | bkw_ | we are doing the test later today |
| 21:55.38 | CyberKnet | could be weeks, or could be months kind of deal? |
| 21:55.47 | bkw_ | no |
| 21:56.25 | ClayReiche123 | I have that line in my INVITE from my sip phone TO asterisk.... when Asterisk send the INVITE to the Sonus, that attribute is missing... |
| 21:56.48 | shido6 | Sonus |
| 21:56.50 | shido6 | oh gawd |
| 21:56.55 | CyberKnet | okay. |
| 21:57.01 | ClayReiche123 | no kidding... |
| 21:57.03 | shido6 | I smacked Sonus a few times when I was a t GX |
| 21:57.18 | shido6 | before they filed for bankruptcy protection |
| 21:57.19 | shido6 | bastards |
| 21:57.24 | blitzrage | PRI <-> Asterisk <-g.729 w/ rfc2833-> IVR the IVR sees duplicate DTMF digits, but I have no idea why, any suggestions? |
| 21:57.27 | ClayReiche123 | Qwest uses it. |
| 21:58.58 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
| 21:58.58 | ClayReiche123 | I think I'm back to wishing I chose SER... I know I could manipulate the INVITE... (probably wouldn't have too though...) |
| 21:59.12 | shido6 | so toss in ser |
| 21:59.21 | shido6 | Sir Asterisk |
| 21:59.22 | ClayReiche123 | I don't know it.... |
| 21:59.25 | ClayReiche123 | at all... |
| 21:59.26 | harryvv | clay you used ser yet? |
| 21:59.32 | shido6 | time to lear |
| 21:59.33 | shido6 | n |
| 21:59.33 | ClayReiche123 | no |
| 21:59.46 | devel | i just updated to the latest cvs (today), and now dtmf (rfc2833) on my grandstream devices (bt and ht) doesn't work to asterisk (vm, transfer) but works extermally, and works fine for other devices (polycom, sipura, snom). is it just me? |
| 21:59.47 | ClayReiche123 | I'm thinking yo're right.... |
| 22:00.00 | CyberKnet | bkw_: please excuse my seeming impatience. I'm just trying to figure out if I need to sign up with another provider in the mean time, or stay with vonage till you have it available. |
| 22:00.19 | harryvv | Cyber, heck sign up with sixtel |
| 22:00.32 | harryvv | if you want long distance |
| 22:00.33 | harryvv | :) |
| 22:01.02 | AgiNamu | I LIKE ASTERISK |
| 22:01.09 | AgiNamu | Sorry... Loud Howard moment there. |
| 22:01.12 | CyberKnet | harryvv: no, I want DID |
| 22:01.34 | harryvv | cyber, sixtel offers did |
| 22:01.45 | ClayReiche123 | I DO TOO.... I just wish I could manipulate INVITES.... :) |
| 22:01.50 | CyberKnet | harryvv: Not in my NPA NXX |
| 22:02.00 | harryvv | cyber, mmm that sucks. |
| 22:02.23 | harryvv | what area code/location |
| 22:02.30 | CyberKnet | teliax has it, but asterlink rocks beyond what teliax can even comprehend =) |
| 22:02.31 | blitzrage | is there a way I can verify that DTMF is being sent via RFC2833 instead of inband? |
| 22:02.34 | CyberKnet | 918-481 |
| 22:02.57 | bkw_ | rtp debug |
| 22:03.02 | blitzrage | bkw_: thanks |
| 22:03.16 | blitzrage | oh yay... guess not in 1.0 |
| 22:03.26 | ClayReiche123 | It falls short for me enough that I wish I knew c and could participate in the development.... |
| 22:07.07 | harryvv | anything ip or tel .com releated is about exausted. Looking for a company name that sound even remotely clost to voip is a long shot. |
| 22:07.42 | CyberKnet | GiveMeYourDamnMoneyOIP |
| 22:08.15 | harryvv | yea, one syber squater wanted $10k for a voip name. |
| 22:08.24 | harryvv | cybersquater |
| 22:08.34 | CyberKnet | hate cybersquatters. |
| 22:08.37 | CyberKnet | parasites. |
| 22:08.43 | blitzrage | why does everyone hate me? :) |
| 22:08.50 | ChkDigit | You mean someone has taken kissmyasstelcoimusingvoip.com ? |
| 22:08.54 | ClayReiche123 | so... no ideas with my INVITE problem? |
| 22:09.05 | marky | i tried to buy interweblog.com from a guy and he wanted $12k |
| 22:09.06 | harryvv | and then there are those almost worhtless web sites that has some domain name that sounds like a telephone,ip,voip or what ever releated. |
| 22:09.19 | marky | and when i laughed it him he said it was negotiable |
| 22:09.37 | marky | but i didn't figure it was negotiable to $50 so i never responded |
| 22:09.51 | CyberKnet | harryvv: well, if you cant beat them... join them. start registering tle and iop .com domaisn =) |
| 22:10.04 | harryvv | There was one company that wanted to sell me northtel.com for 12k and i said I had to pass..it sounded to much like nortel and I could see the confusion with my future company name. |
| 22:10.50 | marky | that's crazy though....that's like ....alot of moolah |
| 22:10.50 | marky | you could probably get taken to court for it as well |
| 22:10.50 | marky | for being a sound alike... |
| 22:10.58 | harryvv | I know |
| 22:11.03 | harryvv | was thinking the same thing. |
| 22:11.07 | denon | way OT: any of you guys know someone who works a lot with photography/lighting? |
| 22:11.24 | marky | if domain names were cheaper...i'd probably do it but at $8 a pop...that's too much |
| 22:11.26 | harryvv | denon, as in cinema photography? |
| 22:11.33 | denon | harryvv: nah, still |
| 22:11.35 | harryvv | or stills |
| 22:11.36 | harryvv | okay |
| 22:11.45 | *** join/#asterisk |Vulture| (~V@199.227.253.212) |
| 22:11.51 | denon | you fit that bill? |
| 22:11.57 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.126) |
| 22:12.11 | harryvv | no, just had a classmate who is a cinamaphotographer. |
| 22:12.12 | blitzrage | I might have an idea for your INVITE problem is you have an idea for my duplicate RFC2833 DTMF digits |
| 22:12.15 | blitzrage | :) |
| 22:12.23 | harryvv | denon i do decent 3d modeling. |
| 22:12.24 | *** join/#asterisk likwid-- (likwid@nc-69-34-122-245.dyn.sprint-hsd.net) |
| 22:12.29 | |Vulture| | Can anyone give me a hand, trying to configure a TE110P and TDM with 2 FXS cards, the PRI works fine but the TDM I cant get to work here is the config/output I am sure its something simple |
| 22:12.29 | |Vulture| | http://pastebin.ca/11702 |
| 22:12.42 | denon | harryvv: well, this is a lighting question .. like real lights, not just perspectives on a screen :) |
| 22:12.52 | harryvv | what about them |
| 22:13.44 | *** part/#asterisk TonyM (~softins@adsl-solo-80-168-226-67.claranet.co.uk) |
| 22:13.46 | denon | (moved to privmsg :) |
| 22:15.40 | *** part/#asterisk OzJames79 (~opera@CPE20320889-1842-1.gex.ncable.net.au) |
| 22:17.13 | |Vulture| | anyone? |
| 22:18.13 | blitzrage | |Vulture|: seems everyone is busy working on other things right now |
| 22:19.10 | AgiNamu | I can use fdprint* on a socket? |
| 22:22.10 | extremis | ok, I figured out my cdr problem, source isn't ever the SIP channel but instead is the callerid that is being set when it leaves asterisk (ie when it leaves the pri) |
| 22:22.21 | extremis | is there any way to make the source show up as the extension? |
| 22:23.08 | *** join/#asterisk likwid-- (likwid@nc-65-41-163-133.dyn.sprint-hsd.net) |
| 22:23.34 | *** join/#asterisk meppl (mephisto@p54AAEDA9.dip.t-dialin.net) |
| 22:23.55 | |Vulture| | well I got it, all it needed was to be moved out of the PRI banking... so 25-26 worked |
| 22:23.58 | devel | |Vulture|, did you try putting the fxo at chan 25-26 (i.e. outside of the span) |
| 22:24.30 | |Vulture| | hahaha nice devel ;) |
| 22:27.59 | *** join/#asterisk juice (~juice@mo-65-41-222-69.dyn.sprint-hsd.net) |
| 22:30.04 | *** join/#asterisk jeffgus (~jeffgus@alpha.zimage.com) |
| 22:30.11 | *** join/#asterisk Legend (~legend@24.244.142.133) |
| 22:30.21 | *** join/#asterisk dalabera (~Dalabera@mail.pmrtechnologies.com) |
| 22:30.27 | dalabera | hello everyone!!! |
| 22:34.11 | |Vulture| | hey |
| 22:38.19 | *** part/#asterisk ClayReiche123 (fwuser@mail.accxx.com) |
| 22:41.22 | pussfeller | how do I hang up the pstn line |
| 22:41.38 | pussfeller | soft hangup various combots dont work |
| 22:43.55 | |Vulture| | like hangup a single zap line? |
| 22:44.00 | pussfeller | yes |
| 22:44.37 | pussfeller | the real question is why didn't it hang up its self after voice mail |
| 22:45.18 | |Vulture| | pussfeller: possibly you don't have disconnect supervision |
| 22:45.35 | |Vulture| | Ive seen it in only 1 set of lines I have ever used |
| 22:45.48 | |Vulture| | in that case I use a timeout... until I can get it fixed |
| 22:46.15 | pussfeller | is that a signal the phone company sends? |
| 22:47.04 | pussfeller | i remember the fiber optic guy teling me the lines took a real long time to hang up here |
| 22:47.59 | |Vulture| | pussfeller: yes thats correct |
| 22:48.31 | |Vulture| | http://www.voip-info.org/wiki-Asterisk+Disconnect+Supervision |
| 22:48.48 | pussfeller | well i did soft hangup Zap/1-1 and that took it off offhook |
| 22:49.00 | pussfeller | but ended up with a 32 minute VM message |
| 22:49.04 | AgiNamu | well, i got my send audio code working |
| 22:49.09 | AgiNamu | err, written |
| 22:49.12 | pussfeller | its set to 3 minutes in voicemail.conf |
| 22:49.13 | AgiNamu | yet to test and try it out |
| 22:49.38 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
| 22:52.20 | *** join/#asterisk L|NUX (~linux@202.5.145.54) |
| 22:53.29 | *** join/#asterisk iamnotbob (~nolan@216.199.159.79) |
| 23:06.38 | *** join/#asterisk ajnewbold (~ajnewbold@12-202-33-32.client.insightBB.com) |
| 23:08.53 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 23:14.17 | ManxPower | Apparently ariel_ talked about his asterisk server again. *tease* |
| 23:14.25 | docelmo | When I download head from cvs I use this flag correct? -r v1-0 |
| 23:14.26 | |Vulture| | lol |
| 23:14.39 | |Vulture| | no thats stable |
| 23:14.40 | ManxPower | docelm0: that downloads CVS 1.0.x |
| 23:14.45 | |Vulture| | yea |
| 23:14.46 | *** join/#asterisk critch (critch@steven.basesys.com) |
| 23:14.50 | |Vulture| | the 1.0 branch |
| 23:14.54 | |Vulture| | not the true HEAD |
| 23:14.57 | ManxPower | He sure is chatty for being AFK |
| 23:15.23 | Legend | anyone ever integrated a * box with an avaya ipoffice? |
| 23:15.25 | |Vulture| | damn those door fones are easy to install |
| 23:15.49 | *** join/#asterisk habakuk (~habakuk@24-116-201-136.cpe.cableone.net) |
| 23:15.50 | docelmo | Whats the flag for HEAD? |
| 23:16.07 | Corydon-w | No flag |
| 23:16.08 | Nugget | there is no flag for HEAD> |
| 23:16.10 | Nugget | it's the default |
| 23:16.15 | docelmo | o |
| 23:16.21 | Corydon-w | That's the whole point of HEAD... it's not a branch |
| 23:16.22 | docelmo | damnit.. I need to download it again then |
| 23:16.35 | Nugget | more importantly, what's the flag to turn off that annoying away message? |
| 23:16.52 | docelmo | huh? |
| 23:16.57 | Nugget | 18:14 <docelm0 is AFK, home.. not at work!. Gone now for:�1day 2hrs 1min� (Pager is On, /ctcp docelm0 Page <message>) WI-n-v-i-s-i-o-nW |
| 23:17.16 | Nugget | turn that shit off. |
| 23:17.37 | docelmo | ohh.. I didnt know it was doing it. I will see if I can kill it.. I will login to my office later tonight and look at it. |
| 23:17.58 | docelmo | its probably one of the timers |
| 23:18.20 | *** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 23:19.49 | habakuk | anyone remember how to allow a sip device only from a certain ip to make calls without authentication? |
| 23:20.59 | *** join/#asterisk outtolunc (~me@adsl-69-110-50-162.dsl.pltn13.pacbell.net) |
| 23:24.18 | ariel_ | habakuk, if its the same as iax.conf it's permit=xxx.xxx.xxx.xxx/255.255.255.0 deny=0.0.0.0/0.0.0.0 |
| 23:24.40 | ariel_ | habakuk, then inscure=yes |
| 23:24.44 | Legend | whats the best h323 channel driver to be using with stable? |
| 23:24.52 | habakuk | arielm thx |
| 23:25.12 | *** join/#asterisk Cassador (cass@bl4-152-221.dsl.telepac.pt) |
| 23:25.23 | Cassador | Salute gents |
| 23:25.24 | ariel_ | Legend, there are 2 oh323 and the one that comes with asterisk h323 there is also a new one that is in testing. |
| 23:25.41 | Legend | ok |
| 23:25.53 | Legend | ariel_: so running -stable i shouldn't need to compile in anything? |
| 23:26.25 | ariel_ | Legend, yes you do see the h323 readme |
| 23:26.39 | Legend | ok, thanks |
| 23:26.53 | *** part/#asterisk ajnewbold (~ajnewbold@12-202-33-32.client.insightBB.com) |
| 23:26.56 | docelmo | Legend go here: |
| 23:27.06 | docelmo | http://lists.digium.com/pipermail/asterisk-users/2005-January/081651.html |
| 23:27.23 | docelmo | its what I am using now to build h323 the god forsaken protocol.. |
| 23:27.39 | Legend | docelmo: nice |
| 23:28.03 | Legend | docelmo: i am going to take a masochistic trip this weekend, we have an avaya ipoffice kicking around, wanna see if i can integrate ;-D |
| 23:28.22 | docelmo | I would imagine it wouldnt be an issue. |
| 23:28.41 | Legend | and you like oh323 better than the built in channel driver |
| 23:28.53 | docelmo | I have worked with SIP so much I prefer it over ANY protocol except IAX it works so well |
| 23:29.05 | docelmo | yes.. I find its less headache to build |
| 23:29.12 | Legend | okie |
| 23:29.20 | docelmo | what OS are you gonna build it on? |
| 23:29.33 | Legend | linux, debian sarge |
| 23:29.36 | docelmo | I have no problems with CentOS but FC3 is kicking my ass |
| 23:29.56 | docelmo | I dont know about debian.. I have used and like it but Im hooked on Cent now |
| 23:30.06 | Legend | ill give it a whirl |
| 23:30.22 | ariel_ | Legend, give this a try http://www.computers4sure.com/product.asp?productid=102876&affid=10001152 |
| 23:30.36 | ariel_ | it's the new driver that there trying to get into asterisk and they need testers. |
| 23:30.42 | Legend | ariel_: yeah, looks good |
| 23:30.47 | Legend | i used that one already |
| 23:30.55 | Legend | ;-P |
| 23:30.55 | docelmo | Memory Issues? |
| 23:31.16 | ariel_ | Legend, this one is going to be supported by digium from the looks of it. |
| 23:31.30 | Legend | ariel_: are you sure you pasted the right link? |
| 23:31.39 | docelmo | ya its for a stick of rqm |
| 23:31.41 | docelmo | err ram |
| 23:31.45 | ariel_ | the one in asterisk h323 is not hard if you know he is looking for the files to be in the root area for installation. |
| 23:32.09 | ariel_ | Legend, sorry http://bugs.digium.com/view.php?id=4234 |
| 23:32.09 | docelmo | well jj shouldnt have coded it that way |
| 23:32.14 | ariel_ | I was doing 2 things |
| 23:32.22 | Legend | ariel_: hehe |
| 23:32.49 | docelmo | That one I have tried.. builds beautiful but lacking lots of codec support and features.. |
| 23:32.57 | ariel_ | trying to get some memory for a netfinity server. |
| 23:32.59 | docelmo | I need something I can use in production now unfortuneatelly |
| 23:33.35 | docelmo | ahh.. |
| 23:33.54 | docelmo | I use all supermicro Xeon and Tyan Opteron hardware platforms |
| 23:34.35 | *** join/#asterisk pussfeller (~todd@t1-rtc-woodlawn.rtcol.com) |
| 23:34.38 | ariel_ | docelmo, it's a customers that needs W2K 2003 SBS installed. It pays the bills service calls |
| 23:35.03 | docelmo | this is ture |
| 23:35.06 | docelmo | err true |
| 23:35.12 | ariel_ | besides supermicro's sata raid is only supporting rh 9 right now. There refusing to upgrade the driver. |
| 23:35.28 | docelmo | with SBS watch workstation licensing.. its a bitch |
| 23:36.11 | ariel_ | docelmo yes your correct. It does suck. But this is an architech office and all they have is XP pro for autocad. |
| 23:38.27 | docelmo | ya.. I had to upgrade all my servers to 2003 standard cause of the licensing shit. It was cheaper that way. I have 150 client machines 25 servers in my main office not to mention VPN and remote sites that use our network |
| 23:38.50 | ariel_ | k |
| 23:40.54 | *** join/#asterisk nords (~chatzilla@S01060012172e2b25.no.shawcable.net) |
| 23:41.37 | ManxPower | Um, so you have 6 clients per server? You must be running Windows on the servers. |
| 23:42.06 | docelmo | yes.. |
| 23:42.13 | ManxPower | Wow? |
| 23:42.15 | docelmo | I have to for alot of the applications we run. |
| 23:42.16 | ManxPower | ..er Wow! |
| 23:42.27 | ManxPower | Why not just run them on the desktops? |
| 23:42.28 | docelmo | out of the 25 I have 10 linux servers |
| 23:42.37 | docelmo | Cant run MAS on the desktop |
| 23:42.45 | docelmo | and others like it |
| 23:42.59 | docelmo | alot of proprietary crap.. Were a telco in Tampa, FL |
| 23:43.02 | *** join/#asterisk CdtDelta (~CdtDelta@dsl081-225-161.chi1.dsl.speakeasy.net) |
| 23:43.03 | nords | hopefully someone can help me out here. For some reason when ever i issue a "redirect" command to a new conference it hangs up the channel. This was working for me in the past |
| 23:43.37 | docelmo | over my head.... |
| 23:43.46 | nextime | anyone using pyastre? |
| 23:44.15 | ManxPower | We managed to get rid of all servers and replace them with Linux, except for the Windows Terminal Server and the AS/400 |
| 23:45.14 | ariel_ | nords, I have not used the redirect but what does it say on the cli when this happens? |
| 23:45.48 | docelmo | I wish.. but Im the IT guy in charge but I dont make those decisions.. |
| 23:46.06 | docelmo | Manx have you guys been able to get Samba to intergrate into a windows domain? |
| 23:46.36 | docelmo | I could with NT/2000 but 2k3 isnt working very well |
| 23:46.45 | nords | Action: Redirect |
| 23:46.47 | nords | Channel: IAX2/test@70.66.92.189:4569-3 |
| 23:46.48 | nords | Extension: 9011 |
| 23:46.50 | nords | Response: Success |
| 23:46.51 | nords | Message: Redirect successful |
| 23:46.53 | nords | Event: MeetmeLeave |
| 23:46.54 | nords | Privilege: call,all |
| 23:46.56 | nords | Channel: IAX2/test@70.66.92.189:4569-3 |
| 23:46.57 | nords | Uniqueid: 1116027810.0 |
| 23:46.59 | nords | Meetme: 9010 |
| 23:47.00 | nords | Usernum: 1 |
| 23:47.02 | nords | Event: Hangup |
| 23:47.04 | nords | Privilege: call,all |
| 23:47.05 | nords | Channel: Zap/pseudo-784886717 |
| 23:47.07 | nords | Uniqueid: 1116027851.1 |
| 23:47.07 | docelmo | DUDE! PASTEBIN.CA |
| 23:47.08 | nords | Cause: 0 |
| 23:47.10 | nords | Event: Hangup |
| 23:47.11 | nords | Privilege: call,all |
| 23:47.13 | nords | Channel: IAX2/test@70.66.92.189:4569-3 |
| 23:47.14 | nords | Uniqueid: 1116027810.0 |
| 23:47.16 | nords | Cause: 0 |
| 23:47.21 | docelmo | or not |
| 23:48.19 | nords | sorry, I didn't think there was enough for pastebin |
| 23:48.28 | nords | should i have used it? |
| 23:48.29 | denon | >4 lines is plenty for a pastebin |
| 23:49.38 | nords | http://pastebin.ca/11709 |
| 23:50.31 | *** join/#asterisk mike-ff (~mike@63.121.58.33) |
| 23:51.33 | nords | the stupid thing is the redirect was working for a couple of months up till today. I have tried it on both the debian stable package, and also CVS Head as of the May 12th |
| 23:52.39 | ariel_ | nords, what did you change today? |
| 23:54.23 | nords | <PROTECTED> |
| 23:54.37 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 23:54.39 | PTG123 | May 12 23:50:56 WARNING[44678]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x815421c (len 440) to 64.201.119.146 returned -1: Can't assign requested address |
| 23:54.42 | PTG123 | anyone seen that before? |
| 23:55.10 | pussfeller | whats the syntax for setting absolut timeout |
| 23:55.17 | pussfeller | i cant find a real example |
| 23:57.21 | *** join/#asterisk NewSole2 (dave@i216-58-44-245.avalonworks.net) |